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检索条件"主题词=Audio coding"

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audio coding using the best level wavelet packet transform and auditory masking

Audio coding using the best level wavelet packet transform a...

引用

International Conference on Signal Processing Proceedings (ICSP)

作者： S. Hosoi J. Toyama M. Shimbo Division of Systems & Information Engineering Graduate School of Engineering Hokkaido University Sapporo Japan

In this paper, we propose an audio coder that aims at high quality even in nonstationary segments of music signals. First, we divide input signals into subbands by using wavelet packet decomposition. We use a critical band approximate wavelet packet decomposition tree for efficient auditory masking. Dynamic wavelet packet decomposition is then used for effective signal representation. By choosing the basis from a wavelet packet tree, the best level method is used. The bases of the best level method has more entropy than that of the best basis method, but by choosing the appropriate frame length, it has similar entropy to that of the bases of the best basis method. Thus we can reduce side information. We propose an adaptive frame method whose criteria is the entropy of best level basis.

关键词： audio coding Wavelet packets Wavelet transforms Entropy Bit rate Quantization Frequency Decoding Signal resolution Tree data structures

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audio coding using parametric piecewise modeling

Audio coding using parametric piecewise modeling

引用

IEEE International Conference on Personal Wireless Communication

作者： M. Imine A.R. Prasad Lucent Technol. Netherlands Wireless Communications & Networking Division Lucent Technologies Inc. Nieuwegein Netherlands

We study the application of parametric piecewise modeling (PPM), a novel coding technique, for audio coding. PPM is based on either the polynomial, Bezier or Chebyshev parametric representation. It has already been proposed for curve and surface modeling use in computer graphics and has been recently proposed for source and channel coding. PPM speeds up the process of generating graphics, has increased compression capability and to a certain extent can predict missing parts of the signal. Results show that high compression of audio signals is achieved using PPM. Further we can see that error correction, to a certain extent, can be achieved using PPM due to its prediction capabilities.

关键词： audio coding Source coding Channel coding Polynomials Computer graphics Signal processing Application software Chebyshev approximation Signal generators Error correction

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audio coding using steady state harmonics and residuals

Audio coding using steady state harmonics and residuals

引用

IEEE Workshop on Multimedia Signal Processing

作者： K.N. Hamdy A.H. Tewfik Department of Electrical and Computer Engineering University of Minnesota Minneapolis MN USA

In this paper, we solve the problem of inter and intra-frame frequency leakage in harmonic based audio coding approaches. Specifically, we first use an FFT to estimate the harmonic parameters. Because of the limited frequency resolution of the windowed FFT, there is an uncertainty in the measured frequency within one frequency bin. To minimize this uncertainty, we use these estimates as seeds to a minimum least squares problem and solve for the optimal harmonic parameters. The resulting residual should be essentially free of harmonics and consist of only attacks and noise. We are preserving the attacks and coding them separately. The resulting residual noise is modeled and coded using wavelet transforms and noise modeling. We also present a solution to the problem of inter-frame frequency fading and highlight the benefit of using a rectangular coordinate representation of the harmonic data. The improved coder that we propose can be viewed as an object based approach to audio coding and leads to perceptually pleasant progressive transmission and representation schemes for multimedia applications.

关键词： audio coding Steady-state Frequency Speech synthesis Signal synthesis Harmonic analysis Speech analysis Discrete wavelet transforms Wavelet transforms Signal analysis

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audio coding With Unified Noise Shaping And Phase Contrast Control

Audio Coding With Unified Noise Shaping And Phase Contrast C...

引用

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

作者： Byeongho Jo Seungkwon Beack Taejin Lee Electronics and Telecommunications Research Institute Republic of Korea

Over the past decade, audio coding technology has seen standardization and the development of many frameworks incorporated with linear predictive coding (LPC). As LPC reduces information in the frequency domain, LP-based frequency-domain noise-shaping (FDNS) was previously proposed. To code transient signals effectively, FDNS with temporal noise shaping (TNS) has emerged. However, these mainly operated in the modified discrete cosine transform domain, which essentially accompanies time domain aliasing. In this paper, a unified noise-shaping (UNS) framework including FDNS and complex LPC-based TNS (CTNS) in the DFT domain is proposed to overcome the aliasing issues. Additionally, a modified polar quantizer with phase contrast control is proposed, which saves phase bits depending on the frequency envelope information. The core coding feasibility at low bit rates is verified through various objective metrics and subjective listening evaluations.

关键词： Measurement audio coding Frequency-domain analysis Computational modeling Discrete Fourier transforms Bit rate Standardization

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audio coding using sinusoidal excitation representation

Audio coding using sinusoidal excitation representation

引用

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

作者： Wen-Whei Chang De-Yu Wang Li-Wei Wang Department of Communication Engineering National Chiao Tung university Hsinchu Taiwan

ISBN: (纸本)0818679190

Most LPC-based audio coders employ simplistic noise-shaping operations to perform psychoacoustic control of quantization noise. In this paper, we report on new approaches to exploiting perceptual masking in the design of adaptive quantization of LPC excitation parameters. Due to its localized spectral sensitivity, sinusoidal excitation representation is preferred to spectrally flat signals for use in excitation modeling. Simulation results indicate that the proposed multisinusoid excited coder can deliver high quality audio reproduction at the rate of 72 kb/s.

关键词： audio coding Linear predictive coding Quantization Psychoacoustic models Psychology Filters Frequency Noise shaping Bit rate Signal processing

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audio coding and electronic distribution of music

Audio coding and electronic distribution of music

引用

International Conference on Web Delivering of Music (WEDELMUSIC)

作者： K. Brandenburg Technische Universität Ilmenau and Fraunhofer AEMT Germany

audio coding originally has been developed as a means for making digital radio possible and for distributing audio via phone lines. The most apparent application today is electronic distribution of music via the Internet. While legal services and copy protection methods have been around for seven years, the public attention and the bulk of actual use is still on unauthorized copying of music files. The talk will introduce the basic techniques for high quality coding of music and discuss some possibilities to reduce the unauthorized copying of music.

关键词： audio coding Filter bank Encoding Decoding Signal analysis Psychoacoustic models Masking threshold Quantization ISO IEC standards

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audio coding using the wavelet packet transform and a combined scalar-vector quantization

Audio coding using the wavelet packet transform and a combin...

引用

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

作者： S. Boland M. Deriche School of Electrical and Electronic Systems Engineering Signal Processing Research Centre Queensland University of Technology Brisbane QLD Australia

This paper investigates a hybrid scalar-vector quantization scheme for coding high quality audio signals. A wavelet packet transform (WPT) is used to decompose the audio signal into frequency bands slightly finer than the critical band divisions. A masking model computation is then used as input to the hybrid quantization scheme, where scalar quantization is used for coding the subbands from 0-5.5 kHz, and vector quantization is used for coding the subbands from 5.5-22 kHz. The performance of the proposed coder is assessed from segmental signal-to-noise ratios (SNR) and the perceived quality for a number of signals. The perceived quality is determined from informal comparisons between the uncoded signals at the original bitrate of 705 kb/s, and the same signals coded with (1) the proposed coder at 80 kb/s, (2) a coder using only scalar quantization at both 128 kb/s and 96 kb/s, and (3) the MPEG layer III coder at 64 kb/s. The comparisons indicate that a very good coder quality is possible with the proposed coder at bitrates of approximately 80 kb/s. This represents a saving of about 16 kb/s over full scalar quantization with a similar quality. Further bitrate reduction with the proposed coder is possible by entropy coding of the scalar quantized transform coefficients and the VQ indices.

关键词： audio coding Wavelet packets Wavelet transforms Bit rate Signal processing Frequency conversion Vector quantization Entropy coding Signal resolution Systems engineering and theory

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audio coding Using Masking-threshold Adapted Perceptual Filter

Audio Coding Using Masking-threshold Adapted Perceptual Filt...

引用

IEEE Workshop on Speech coding For Telecommunications

作者： Wen-Whei Chang Chin-Tung Wang Department of Communication Engineering National Chiao Tung University Hsinchu Taiwan

Most current work in LPC-based coding employs either postfitering or noise spectral shaping to take into account the auditory masking effect. In both cases, the LPC parameters are used to characterize both the synthesis filter and the perceptual weighting filter. Since they are originally derived to model the vocal tract, LPC parameters can not precisely describe the nonlinear sensitivity of human car to frequency and to loudness. In this paper, we explore the benefits of masking-threshold adapted perceptual filter for excitation search in analysis-by-synthesis predictive coders. Furthermore, one derivation of Bark spectral distortion is also studied as the objective measure for assessng audio quality.

关键词： audio coding Filters Noise shaping Frequency Masking threshold Linear predictive coding Humans Psychoacoustic models Testing Acoustic noise

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audio coding using a psychoacoustic pre- and post-filter

Audio coding using a psychoacoustic pre- and post-filter

引用

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

作者： B. Edler G. Schuller Multimedia Communications Research Laboratory Bell Laboratories Lucent Technologies Inc. Murray Hill NJ USA

A novel concept for perceptual audio coding is presented which is based on the combination of a pre- and post-filter, controlled by a psychoacoustic model, with a transform coding scheme. This paradigm allows modeling of the temporal and spectral shape of the masked threshold with a resolution independent of the used transform. By using frequency warping techniques the maximum possible detail for a given filter order can be made frequency-dependent and thus better adapted to the human auditory system. The filter coefficients are represented efficiently by LSF parameters which can be adaptively interpolated over time. First experiments with a system obtained by extending an existing transform codec showed that this approach can significantly improve the performance for speech signals, while the performance for other signals remained the same.

关键词： audio coding Psychology Psychoacoustic models Frequency Transform coding Spectral shape Humans Auditory system Adaptive filters Codecs

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audio coding using sorted sinusoidal parameters

Audio coding using sorted sinusoidal parameters

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： M. Raad I.S. Burnett School of Electrical Computer and Telecommunications Engineering University of Wollongong Wollongong NSW Australia

ISBN: (纸本)0780366859

This paper describes a new audio coding scheme based on sinusoidal coding of signals. Sinusoidal coding permits the representation of a given signal through the summation of sinusoids. The parameters of the sinusoids (the amplitudes, phases and frequencies) are transmitted to allow the signal reconstruction. In the proposed scheme, the sinusoidal parameters are sorted according to energy content and perceptual significance. The most significant parameters are transmitted first allowing the use of only a small set of the parameters for signal reconstruction. The proposed scheme incurs a low delay and uses a 20 ms frame length. Results show that the coder operating at a mean rate of 39 kb/s, performs favorably in comparison with the MPEG-4 coder at 42 kb/s.

关键词： audio coding Frequency Bit rate Signal reconstruction Sorting Fourier transforms Signal synthesis Australia Delay MPEG 4 Standard

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