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检索条件"主题词=Audio coding"

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Investigation on objective performance of closed-loop spatial audio coding

Investigation on objective performance of closed-loop spatia...

引用

International Conference on Information Technology and Electrical Engineering (ICITEE)

作者： Ikhwana Elfitri Rahmadi Kurnia Fitrilina Department of Electrical Engineering Andalas University Kampus Limau Manis Padang 25163 Indonesia

ISBN: (纸本)9781479953042

Closed-loop spatial audio coding is a compression technique, developed based on MPEG Surround (MPS) standard, having an advantage of minimising distortion due to quantisation process of spatial parameters. Despite the MPS is developed based on filterbank, however, this closed-loop system performs better on Modified Discrete Cosine Transform (MDCT). Considering its high performance over the open-loop system, this paper presents further investigation on objective performance of closed-loop spatial audio coding against various quantisers of spatial parameters. Experiments have been conducted to measure signal to noise ratio (SNR) across different types of uniform spatial quantisers at various operating bitrates. The results show that the SNR achieved by the open-loop approach is strongly affected by the type of the quantiser while, in contrast, the SNR achieved by the closed-loop approach is relatively constant regardless the number of bits used in the quantisers. Moreover, the results also show that the closed-loop configuration can consistently improve SNR in any quantisation scheme.

关键词： Bit rate Codecs audio coding Transform coding Signal to noise ratio Electrical engineering

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Spectral amplitude warping (SAW) for noise spectrum shaping in audio coding

Spectral amplitude warping (SAW) for noise spectrum shaping ...

引用

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

作者： R. Lefebvre C. Laflamme University of Sherbrooke QUE Canada

In this paper, we present a new approach to shape the coding noise in speech and audio coders. The approach, called spectral amplitude warping (SAW), consists essentially of a pre- and post-processing which apply a nonlinear transformation to the signal short-term spectrum prior to, and after, encoding. Since it is possible to view SAW as a separate entity from the coder, the noise shaping capability of an existing coder can be improved without modifying the coder itself. Using SAW as a pre- and post-process to the G.722 wideband speech coding standard, it was found in an informal listening test that the quality of the 64 kb/s operating mode can be achieved at only 48 kb/s. The price to be paid is an additional delay.

关键词： Surface acoustic waves Noise shaping Noise level audio coding Speech enhancement Speech coding Encoding Speech synthesis Nonlinear filters Wideband

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Extended linear prediction tools for lossless audio coding

Extended linear prediction tools for lossless audio coding

引用

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

作者： T. Moriya D.T. Yang T. Liebchen NTT Cyber Space Laboratories Tokyo Japan University of Southern California Los Angeles USA Technical University Berlin Berlin Germany

Two extension tools for enhancing the compression performance of prediction-based lossless audio coding are proposed. One is progressive-order prediction of the starting samples at the random access points, where the information of previous samples is not available. The first sample is coded as is, the second is predicted by first-order prediction, the third is predicted by second-order prediction, and so on. This can be efficiently carried out with PAR-COR (PARtial autoCORrelation) coefficients. The second tool is interchannel joint coding. Both predictive coefficients and prediction error signals are efficiently coded by interchannel differential or three-tap adaptive prediction. These new prediction tools lead to a steady reduction in bit rate when random access is activated and the interchannel correlation is strong.

关键词： audio coding Linear predictive coding Decoding Performance loss Bit rate Entropy Autocorrelation Multiple signal classification Propagation losses MPEG standards

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Scalefactor based bit shift FGS audio coding

Scalefactor based bit shift FGS audio coding

引用

International Conference on Advanced Information Networking and Applications (AINA)

作者： Fang-Chu Chen Te-Ming Chiu Computer & Communications Research Laboratories Industrial Technology and Research Institute Taiwan

ISBN: (纸本)0769522491

In this paper, a method for applying psychoacoustic effects into the reformation of an MPEG-4 bit sliced arithmetic coded (BSAC) scalable audio bitstream is presented. The method is to rearrange the compressed information based on the scalefactors calculated by the psychoacoustics model in order to reflect the different subjective significance of the compressed data. At low bit rate it improves the coding efficiency by a considerable amount.

关键词： audio coding Psychoacoustic models Bit rate MPEG 4 Standard Streaming media Quantization Psychology Video coding Humans Ear

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High quality audio coding using analysis-by-synthesis technique

High quality audio coding using analysis-by-synthesis techni...

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

作者： X. Lin R.A. Salami R. Steele Department of Electronics and Computer Science University of Southampton Southampton UK Communication Research Center University of Sherbrooke Sherbrooke QUE Canada

A modified regular-pulse excitation (RPE) linear predictive coding (LPC) codec is used to encode audio signals of 15 kHz bandwidth. With appropriate perceptual masking and carefully selected excitation structure, a fixed-pulse excitation (FPE) LPC codec is able to provide high-fidelity audio output. The quantization of both the LPC parameters and the excitation pulse sequence is examined. It is found that the LPC parameters can be quantized efficiently by adaptive differential PCM, while the excitation pulse sequences need more bits to quantize the amplitudes compared to narrowband speech encoding. Informal listening tests show that high-fidelity audio can be achieved at bit rates less than 100 kb/s.< >

关键词： audio coding Linear predictive coding Codecs Bandwidth Quantization Phase change materials Narrowband Speech Encoding Testing

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Perceptual audio coding Using N-Channel Lattice Vector Quantization

Perceptual Audio Coding Using N-Channel Lattice Vector Quant...

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

作者： J. Ostergaard O. Niamut J. Jensen R. Heusdens Information and Communication Theory Group Delft University of Technnology Delft Netherlands

We consider the problem of reliable distribution of audio over packet-switched networks. We make use of multiple-description coding combined with transform coding in order to obtain robustness towards packet losses. Previous approaches to this problem were restricted to the case of only two descriptions. In this work we use n-channel multiple-description lattice vector quantizers (MD-LVQs), which allow for the possibility of using more than two descriptions. For a given packet-loss probability we find the number of descriptions and the bit allocation between transform coefficients which minimizes a perceptual distortion measure subject to an entropy constraint. The optimal quantizers are presented in closed form, thus avoiding any iterative quantizer design procedures. The theoretical results are verified with numerical computer simulations using audio signals and it is shown that in environments with excessive packet losses it is advantageous to use more than two descriptions. We verify in subjective listening tests that using more than two descriptions lead to signals of perceptually higher quality

关键词： audio coding Lattices Vector quantization Distortion measurement Transform coding Robustness Bit rate Entropy Computer simulation Testing

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Acoustic Model Adaptation on Speech and audio coding Distortion

Acoustic Model Adaptation on Speech and Audio Coding Distort...

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Speech Communication;10. ITG Symposium

作者： Ivan Kraljevsk Frank Duckhor Ruediger Hoffmann Institute of Acoustics and Speech Communication TU Dresden

ISBN: (纸本)9783800734559

This paper studies the influence of coding distortion on speech recognition performance. The effects of the coding-decoding process on the Gaussian distributions were investigated by statistics comparison of clean and coded speech data. Further, several different coders were used for speech recognition performance assessment of the UASR (Unified Automatic Speech Recognition and Synthesis) system. The assessment was performed firstly on acoustic model trained on clean speech and latter on coder adapted acoustic models.

关键词： Acoustic models Speech recognition Speech Gaussian distribution ENCODER audio coding abnormal shapes distortion

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A progressive approach for perceptual audio coding

A progressive approach for perceptual audio coding

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IEEE International Conference on Multimedia and Expo (ICME)

作者： Ye Shen Hongmei Ai C.-C.J. Kuo Integrated Media Systems Center Department of Electrical Engineering4 ystem University of Southern California Los Angeles CA USA

A progressive approach to perceptual audio coding is presented in this paper. In contrast with existing perceptual audio coding schemes where the digital audio signal is encoded at a fixed bit rate, the progressive approach achieves fully embedded audio coding whose bit rate can be controlled dynamically, making it useful in audio networking applications. In the proposed scheme, efficient successive quantization and bit layer coding are applied for the compression of an audio signal representation obtained from traditional subband decomposition. An improved successive approximation quantization (ISAQ) is introduced to perform bit allocation and quantization based on the psychoacoustic analysis of input audio signals.

关键词： audio coding Quantization Bit rate Psychology Codecs Image coding Signal analysis Performance analysis Transform coding Humans

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Frequency warping in low delay audio coding

Frequency warping in low delay audio coding

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

作者： S. Wabnik G. Schuller U. Kramer J. Hirschfeld Fraunhofer Institute for Digital Media Technology Ilmenau Germany

The goal of the presented schemes is to obtain an ultra low delay audio coder with a good performance even at low bit rates (around 64 kb/s). The problem to be solved is to gain sufficient frequency resolution at low frequencies for precise low frequency psycho-acoustics and quantization noise shaping, because the ear has a higher frequency resolution at lower frequencies. Our approach is to use warped linear noise shaping pre- and post-filters, and a short DFT for the psycho-acoustic model (length 256), but with frequency warping. We compare four different psychoacoustic versions: DFT with no warping; DFF without warping using warped pre- and post-filters; warping with the so-called NDFT (WDFT); DFT with all-pass delay chain pre-processing. Listening tests show that the best performance is obtained using WDFT.

关键词： Frequency Delay audio coding Psychology Noise shaping Bit rate Quantization Ear Psychoacoustic models Testing

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A wideband speech and audio coding candidate for ITU-T G.711WBE standardization

A wideband speech and audio coding candidate for ITU-T G.711...

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

作者： Yusuke Hiwasaki Takeshi Mori Shigeaki Sasaki Hitoshi Ohmuro Akitoshi Kataoka NTT Cyber Space Laboratories Musashino Tokyo Japan

This paper describes an ITU-T G.711 embedded wideband speech coder, submitted as a candidate to the ITU-T G.711 wideband extension standardization qualification phase. The codec generates a bitstream comprised of three layers: a G.711 core layer with noise shaping, a time-domain weighted vector quantized narrowband enhancement layer, and an MDCT-based higher band enhancement layer. Through subjective evaluations, the coder was found to meet all tested requirements and objectives set in terms of reference, with a low computational complexity at 10 WMOPS.

关键词： Wideband Speech coding audio coding Standardization Qualifications Codecs Noise shaping Time domain analysis Narrowband Testing

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