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检索条件"主题词=Audio coding"

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Fast encoding algorithms for MPEG-4 TwinVQ audio tool

Fast encoding algorithms for MPEG-4 TwinVQ audio tool

引用

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

作者： N. Iwakami T. Moriya A. Jin T. Mori K. Chikira NTT Laboratories Musashino Tokyo Japan

The ISO/IEC MPEG-4 audio standard includes the TwinVQ encoding tool. This tool is suitable for low-bit-rate general audio coding, but a drawback is the computational complexity of the encoder. To develop a faster TwinVQ encoder, new fast vector quantization algorithms - area localized pre-selection and hit zone masking - are introduced. These algorithms exploit the pre- and main-selection procedure scheme of the conjugate structure vector quantization which is used in the TwinVQ. The improvement is evaluated by measuring the encoding speed and the sound quality of reproduction.

关键词： Encoding MPEG 4 Standard Distortion measurement Vector quantization ISO standards IEC standards audio coding Decoding Computational complexity Laboratories

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Automatic parameter optimization for a perceptual audio codec

Automatic parameter optimization for a perceptual audio code...

引用

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

作者： Martin Holters Udo Zolzer Department of Signal Processing and Communications Helmut Schmidt University-University of the Federal Armed Forces Hamburg Hamburg Germany

ISBN: (纸本)9781424423538

In audio codec design, often various parameters have to be fixed which may have a dramatic impact on codec performance. In this paper, we report on successful optimization of a codec based on perceptual criteria. Specifically, the PEAQ measure is used to determine the audio quality over a set of test items and search algorithms are used for optimization. First, simulated annealing is used for global search, then Rosenbrock's method is used to further refine the result. As shown by an example, the improvements gained by optimization compared to an educated guess are substantial.

关键词： Codecs Nonlinear filters Simulated annealing Signal design audio coding Optimization methods Pulse modulation Noise shaping Probability distribution Design optimization

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Over-sampling for accurate masking thershold calculation wavelet packet audio coders

Over-sampling for accurate masking thershold calculation wav...

引用

International Conference on Communication Systems

作者： F. Sinaga E. Ambikairajah A.P. Bradley School of Electrical Engineering and Telecommunications University of New South Wales NSW Australia Cooperative Research Center for Sensor Signal and Information Processing (CSSIP) School of Information Technology and Electrical Engineering University of Queensland QLD Australia

Many existing audio coders use a critically sampled discrete wavelet transform (DWT) for the decomposition of audio signals. While the aliasing present in the wavelet coefficients is cancelled in the decoder, these coders normally perform calculation of the simultaneous masking threshold directly on these aliased coefficients. This paper uses over-sampling in the wavelet packet decomposition in order to provide alias-free coefficients for accurate simultaneous masking threshold calculation. The proposed technique is compared with masking threshold calculation based upon the FFT and critically-sampled wavelet coefficients, and the results show that a bit rate saving of up to 16 kbit/s can be achieved using over-sampling

关键词： Wavelet packets Discrete wavelet transforms Wavelet coefficients Masking threshold Filter bank Sampling methods Australia Signal processing Decoding audio coding

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An overview of MDCT for Time Domain Aliasing Cancellation

An overview of MDCT for Time Domain Aliasing Cancellation

引用

International Conference on Communication and Network Technologies (ICCNT)

作者： V. Arun Raj M. Davidson Kamala Dhas D. Gnanadurai Mepco Schlenk Engineering College Sivakasi India Department of ECE Mepco Schlenk Engineering College Sivakasi India J.P. College of Engineering Ayikudi India

Modified Discrete Cosine Transform (MDCT) is a upcoming special transform implemented in areas of audio signal processing and compression. As, the name implies it is the modified form of DCT (Discrete Cosine Transform) that allows overlapping of segments (say 50%) and thereby it helps in avoiding artifacts. MDCT in general is a lapped transform that rectifies the problem of TDAC (Time Domain Aliasing Cancellation) and hence widely used in audio codec (.mp3, .wav) applications. The scope of this paper is to use MDCT in a system that overcomes the problem of TDAC and implementation of different windows those are capable with this transform. Finally, we calculate the MSE (Mean Squared Error) for the system with different inputs stating that perfect audio reconstruction is possible.

关键词： Time-domain analysis Discrete cosine transforms Speech Speech processing audio coding

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Flexible frequency decompositions for cosine-modulated filter banks

Flexible frequency decompositions for cosine-modulated filte...

引用

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

作者： O.A. Niamut R. Heusdens Department of Mediamatics Delft University of Technnology Delft Netherlands

We investigate the use of nonuniform cosine-modulated filter banks for audio coding. A rate-distortion framework is employed, similar to the work in Herley et al. (1994), to select the filter bank structure from a large library of possible frequency decompositions. A new flexible frequency decomposition algorithm is proposed that jointly optimizes the filter bank structure and the bit allocation over the subband channels. Experimental results for both synthetic and real audio signals are provided. The new algorithm shows significant improvements in comparison with fixed uniform frequency decompositions, but special care has to be taken to reduce the size of the decomposition overhead.

关键词： Channel bank filters Filter bank audio coding Time frequency analysis Wavelet packets Libraries Bit rate Rate-distortion Radio spectrum management Electronic mail

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Measurement and Analysis of Just Noticeable Difference of Interaural Level Difference Cue

Measurement and Analysis of Just Noticeable Difference of In...

引用

International Conference on Multimedia Technology

作者： Weiping Tu Ruimin Hu Heng Wang Wenqin Chen National Engineering Research Center of Multimedia Software Wuhan University of China Wuhan China Computer Science and Electronics department Mondragon Unibertsitatea Mondragon Spain National Engineering Research Center for Multimedia Software Wuhan University Wuhan Hubei CN

Just Noticeable Difference (JND) of interaural level difference (ILD) are measured in 30 subjects with nomal hearing. Single-frequency sine tones with the frequency of 200, 500, 1000, 2000, 5000, 8000 or 12000Hz are used in the subjective listening test. The reference ILDs are 0, 3, 6, 9, 12 and 22dB, indicating 6 directions of sound sources from midline to the direction of the right ear. The results show that direction of the sound source has notable effects on JNDs of ILD. At high frequency, JNDs increase dramatically with source frequency.

关键词： System-on-a-chip audio coding Quantization Transform coding Auditory system Humans Ear

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Modelling the detectability of changes in auditory signals

Modelling the detectability of changes in auditory signals

引用

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

作者： R.E.P. Dowling L.F. Turner Technology and Medicine Imperial College for Science London UK Imperial Coll. of Sci. Technol. & Med. London UK

A model for determining the detectability of changes to auditory signals in monaural two interval listening experiments is proposed. The sensation associated with each interval is modeled as a log excitation vector corrupted by noise. The decision process in the model is based on an optimum unweighted sum of the difference in the log excitations for the two intervals. The performance of the model is assessed using results from psychoacoustical experiments involving single component and multiple component target detection in broadband noise with and without uncertainty as to target frequency. The model is also used to predict results from experiments involving intensity discrimination, variable bandwidth maskers and masker addition. The application of the model to wideband audio coding is discussed.< >

关键词： Psychoacoustic models Wideband Psychology Narrowband Medical signal detection Predictive models audio coding Frequency estimation Biomembranes Educational institutions

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Temporal masking for bit-rate reduction in audio codec based on Frequency Domain Linear Prediction

Temporal masking for bit-rate reduction in audio codec based...

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

作者： Sriram Ganapathy Petr Motlicek Hynek Hermansky Harinath Garudadri IDIAP Research Institute Martigny Switzerland École Polytechnique Fédérale de Lausanne Switzerland Qualcomm Inc. San Diego CA USA

audio coding based on frequency domain linear prediction (FDLP) uses auto-regressive model to approximate Hilbert envelopes in frequency sub-bands for relatively long temporal segments. Although the basic technique achieves good quality of the reconstructed signal, there is a need for improving the coding efficiency. In this paper, we present a novel method for the application of temporal masking to reduce the bit-rate in a FDLP based codec. Temporal masking refers to the hearing phenomenon, where the exposure to a sound reduces response to following sounds for a certain period of time (up to 200 ms). In the proposed version of the codec, a first order forward masking model of the human ear is implemented and informal listening experiments using additive white noise are performed to obtain the exact noise masking thresholds. Subsequently, this masking model is employed in encoding the sub- band FDLP carrier signal. Application of the temporal masking in the FDLP codec results in a bit-rate reduction of about 10% without degrading the quality. Performance evaluation is done with perceptual evaluation of audio quality (PEAQ) scores and with subjective listening tests.

关键词： Codecs Frequency domain analysis Additive white noise audio coding Predictive models Auditory system Humans Ear Acoustic noise Masking threshold

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A new optimized algorithm for computation of MDCT and its inverse transform

A new optimized algorithm for computation of MDCT and its in...

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

作者： Hi-Seok Kim Yang-Ki Cho Won-Pyo Lee Schools of Electronics and Information Engineering Chongju University Cheonghu South Korea Department of Electronic Engineering Chongju University Cheonghu South Korea

ISBN: (纸本)0780386396

The paper presents an optimized algorithm for the computation of the modified discrete cosine transform (MDCT) and its inverse transform (IMDCT). The proposed algorithm is based on the type II DCT (DCT-II). By extracting a common transform kernel to compute both the MDCT and IMDCT, we obtain a unified structure for computation of both of them. Also, since the proposed unified structure is very symmetrical, it is suited to the parallel or pipelined VLSI implementation of MDCT/IMDCT. Furthermore, it requires less calculation complexity than existing algorithms.

关键词： Equations Discrete cosine transforms Discrete transforms Kernel Very large scale integration Filter bank Time domain analysis Digital audio players audio coding Concurrent computing

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Optimization of Masking Expansion Algorithm in Psychoacoustic Models

Optimization of Masking Expansion Algorithm in Psychoacousti...

引用

International Symposium on Intelligence Information Processing and Trusted Computing (IPTC)

作者： Liu Hong-fu Zhang Cong Liang Rui-fan Department of Computer and Information Engineering Wuhan Polytechnic University Wuhan China

MPEG-4 AAC audio coding is the most widely used audio coding at present, but the MPEG-4 AAC audio coding standard has high complexity, long time delay and huge computation, what's more, it is not beneficial for real-time applications. Psychoacoustic model is the core part of the audio encoder, so huge computation also exists. Through researching the masking expansion feature of psychoacoustic, the computation process of the spread function is improved, the psychoacoustic computation decreased, the coding time is reduced, and the experimental results are given, which has important practical meaning to the research of real-time audio coding.

关键词： Psychoacoustic models audio coding Humans Testing Transform coding

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