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检索条件"主题词=Audio coding"

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Fixed bit rate perceptual wavelet packet audio coder

Fixed bit rate perceptual wavelet packet audio coder

引用

International Conference on Communication Systems

作者： T.S. Gunawan E. Ambikairajah J. Epps School of Electrical Engineering and Telecommunications University of New South Wales NSW Australia NICTA Alexandria NSW Australia

This paper presents a perceptual wavelet packet audio coder incorporating temporal and simultaneous masking models with a scalable bit rate allocation scheme. It is demonstrated that the use of even a relatively simple functional model of temporal masking produces bit rate savings in addition to those given by simultaneous masking. The performance of the coder is evaluated using the PEAQ (ITU-R BS.1387) objective measurement criterion for various audio materials given a fixed data rate constraint. Two methods are evaluated to optimally allocate bits, lower band selection, and signal to mask ratio (SMR) selection. Results reveal that for lower bit rates, the lower band selection method provides higher audio quality than that given by SMR selection

关键词： Bit rate Wavelet packets Delay Psychoacoustic models Filter bank Australia Discrete wavelet transforms audio coding Image coding Discrete transforms

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A sparse solution to the bounded subset selection problem: a network flow model approach

A sparse solution to the bounded subset selection problem: a...

引用

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

作者： M. Alghoniemy A.H. Tewfik Electrical Engineering Department Alexandria University Alexandria Egypt Electrical Engineering Department University of Minnesota Minneapolis MN USA

We reformulate the problem of finding the sparsest representation of a given signal using an overcomplete dictionary as a bounded error subset selection problem. Specifically, the reconstructed signal is allowed to differ from the original signal by a bounded error. We argue that this bounded error formulation is natural in many applications, such as coding. Our novel formulation guarantees the sparsest solution to the bounded error subset selection problem by minimizing the number of nonzero coefficients in the solution vector. We show that this solution can be computed by finding the minimum cost flow path of an equivalent network. Integer programming is adopted to find the solution.

关键词： Dictionaries Matching pursuit algorithms Costs Sparse matrices Speech coding audio coding Computer networks Linear programming Signal processing Space power stations

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A statistics study of the MDCT coefficient distribution for audio

A statistics study of the MDCT coefficient distribution for ...

引用

IEEE International Conference on Multimedia and Expo (ICME)

作者： Rongshan Yu Xiao Lin S. Rahardja C.C. Ko Institute for Infocomm Research Singapore Department of Electrical and Computer Engineering National University of Singapore Singapore

The modified discrete cosine transform (MDCT) has been widely used in many transform audio coding algorithms such as MPEG-1/2 layer III (mp3), MPEG-2/4 AAC, Dolby AC2/AC3, and numerous experimental audio coding algorithms. In this paper, we study the probabilistic distribution properties of the MDCT coefficient for audio signals. It is shown that the generalized Gaussian function with distribution parameter r=0.5 or r=1 (Laplacian) provides a good approximation to the distributions of MDCT coefficients for a variety of audio signals. Results from our study also show that although the distribution of these coefficients is not strictly Laplacian, the divergence between them is in fact very small. Therefore, it leads to only marginal redundancy if these coefficients are simply coded with some low-complexity codes designed for Laplacian sources

关键词： Statistical distributions audio coding Filter bank Laplace equations Discrete cosine transforms Entropy Nonlinear filters Discrete transforms Image reconstruction Quantization

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APCC/MDMC '04. The 2004 Joint Conference of the 10th Asia-Pacific Conference on Communications and the 5th International Symposium on Multi-Dimensional Mobile Communications Proceeding

APCC/MDMC '04. The 2004 Joint Conference of the 10th Asia-Pa...

引用

Asia-Pacific Conference on Communications

The following topics are dealt with: TDD-WCDMA MAC protocol; software defined radio; interference suppression; wireless LAN; MIMO systems; MIMO-OFDM communication; space-time modulation; orthogonal code hopping multiplexing; wireless mesh networks; multicast communication; IP networks; session initiated protocol; mobile multimedia MAN; VBLAST systems; smart antenna system; UWB system; satellite ATM; terrestrial networks; phase shift keying; source routing; content delivery server; parallel downloading; heterogeneous networks; fast fading channels; antenna arrays; spatial multiplexing system; audio coding; space-time multiuser detection scheme; blind channel estimation; photonic burst networks; optical transmission system; pervasive computing; DS-CDMA system; GSM system; packet switched networks; differentiated services network; WDM systems; wireless terminals; microstrip filters; QoS architecture; adaptive priority queuing; VoIP networks; watermark detection scheme; dipole array; image sequences; video streaming service; GPRS network; wireless ad hoc network; electromagnetic wave propagation; slab waveguides; mismatched filtering; MPLS network; conformance testing; space-time codes; mobile IP network management; and ad hoc routing protocol.

关键词： Time division multiplexing Code division multiaccess Access protocols Interference suppression Wireless LAN MIMO systems Metropolitan area networks Multimedia communication Adaptive arrays Satellite communication Phase shift keying audio coding Spread spectrum communication Land mobile radio cellular systems Wavelength division multiplexing Microstrip filters Image sequence analysis Electromagnetic propagation Waveguides Communication system operations and management

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A lossless audio compression scheme with random access property

A lossless audio compression scheme with random access prope...

引用

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

作者： Dai Yang T. Moriya T. Liebchen Department of Electrical Engineering University of Southern California Los Angeles CA USA Media Processing Project NTT Cyber Space Laboratory Musashino Tokyo Japan Communication Systems Group Technical University Berlin Berlin Germany

We propose an efficient lossless coding algorithm that not only handles both PCM format data and IEEE floating-point format data, but also provides end users with a random access property. In the worst-case scenario, where the proposed algorithm was applied to artificially generated full 32 bit floating-point sound files with 48 kHz or 96 kHz sampling frequencies, an average compression rate of more than 1.5 and 1.7, respectively, was still achieved, which is much better than the average compression rate of less than 1.1 achieved by the general purpose lossless coding algorithm, gzip. Moreover, input sound files with samples' magnitudes out-of-range can also be perfectly reconstructed by our algorithm.

关键词： audio compression Phase change materials Transform coding audio coding ISO standards USA Councils Laboratories Random media Sampling methods Frequency

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Multiband amplitude modulated sinusoidal audio modeling

Multiband amplitude modulated sinusoidal audio modeling

引用

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

作者： M.G. Christensen S. van de Par S.H. Jensen S.V. Andersen Dept. of Communication Technology Aalborg University Denmark Philips Research Laboratories Eindhoven The Netherlands

In this paper, we investigate the importance of taking frequency-dependent temporal phenomena into account in audio coding. We do this in the context of sinusoidal modeling of audio signals by applying amplitude modulation to the sinusoidal components. Traditionally, audio coders use a fixed time-segmentation for all frequencies despite the fact that it is well-known that the time-frequency resolution of the human auditory system is not constant. The well-known window switching is an example of this. We compare multiband amplitude modulated sinusoidal models to a singleband model using different audio excerpts. Based on both comparative listening tests and a psychoacoustical distortion measure it is concluded that an improvement is generally gained using multiband amplitude modulation, although specific single sources are well-modeled using a singleband model.

关键词： Amplitude modulation Psychoacoustic models Distortion measurement audio coding Context modeling Time frequency analysis Signal resolution Humans Auditory system Testing

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Multi-variate block polar quantization and an application to audio

Multi-variate block polar quantization and an application to...

引用

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

作者： H. Pobloth R. Vafin W.B. Kleijn Department of Signals Sensors and Systems KTH Royal Institute of Technology Stockholm Sweden

We introduce multi-variate block polar quantization (MBPQ). MBPQ minimizes a weighted distortion for a set of complex variables representing one block of a signal under a resolution constraint for the entire block. MBPQ allows for different probability distributions in different dimensions of the set of complex variables. It outperforms a block polar quantizer introduced earlier (Pobloth, H. et al., Proc. Eurospeech, p.1097-100, 2003), and unrestricted polar quantization (UPQ), for both Gaussian complex variables and sinusoids found from audio data. In the case of audio data, we found a performance gain of about 2.5 dB over the best performing conventional resolution-constrained polar quantization, UPQ.

关键词： Quantization Distortion measurement Signal resolution audio coding Speech coding Phase distortion Sensor systems Probability distribution Performance gain Probability density function

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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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Unified architectures for computing DCT/DST/MDCT and their inverses

Unified architectures for computing DCT/DST/MDCT and their i...

引用

IEEE Region 10 International Conference TENCON

作者： M.V. Ramana B. Ray S. Chakrabarti E&ECE Department Kharagpur India IIT G. S. Sanyal School of Telecommunications Kharagpur India

This paper proposes a unified architecture for computation of different Type I/II/III/TV discrete cosine/sine transforms (DCT/DST), modified discrete cosine transform (MDCT) and their inverses. Two methods are described, a parallel one suitable for any length of transform (N) value from 2 to 1024 and a serial one, that uses only one CORDIC up to a maximum length of 32.

关键词： Computer architecture Discrete cosine transforms Telecommunication computing Discrete transforms Hardware Concurrent computing audio coding MPEG 4 Standard Digital audio players Data compression

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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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