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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

Workshop and Exhibition on Moving Picture Experts Group (MPEG-4)

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

ISBN: (纸本)0780371658

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. Finally, additionally optimized encoders are developed by making use of the 3DNow!/SSE technologies. In total, encoding times are reduced to 1/2-1/3 compared to the original TwinVQ encoder in the ISO/IEC reference encoder.

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

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Wavelet based pitch extraction in the mpeg compressed domain

Wavelet based pitch extraction in the mpeg compressed domain

引用

IEEE International Conference on Multimedia and Expo (ICME)

作者： B. Anantharaman K.R. Ramakrishnan S.H. Srinivasan Department of Electrical Engineering Indian Institute of Science Bangalore India

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High-quality and processor-efficient implementation of an MPEG-2 AAC encoder

High-quality and processor-efficient implementation of an MP...

引用

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

作者： Y. Takamizawa T. Nomura M. Ikekawa NEC Corporation Limited Japan

Presents MPEG-2 AAC LC Profile encoder software for an Intel Pentium III processor. Modified discrete cosine transform (MDCT) and quantization processing are accelerated by the use of SIMD instructions. Psycho-acoustic analysis in the MDCT domain makes the use of FFTs unnecessary. Better sound quality is provided by greater efficiency in quantization processing and Huffman coding. All of this results in high-quality and processor-efficient implementation of an MPEG-2 AAC encoder. Sound quality achieved at 96 kbps/stereo is significantly better than that of MP3 at the same bitrate. The encoder works 13 times faster than realtime for stereo encoding on an 800MHz Pentium III processor.

关键词： Encoding Quantization Psychology Digital audio players Software quality Bit rate audio coding Frequency Digital signal processing Microprocessors

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Robust matching of audio signals using spectral flatness features

Robust matching of audio signals using spectral flatness fea...

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： J. Herre E. Allamanche O. Hellmuth Audio Department Fraunhofer Institute for Integrated Circuits Erlangen Germany

Stimulated by the ever-increasing amount of available multimedia data, content-related techniques for the management of audio material have received much interest recently. This paper discusses the problem of robust identification of audio signals by matching them to a known reference. In order to perform well under realworld conditions, the matching process needs to rely on features which are robust with respect to common signal distortions. A family of suitable features with favorable properties is proposed and evaluated for their recognition performance. Applications of signal matching, including fingerprinting, are discussed.

关键词： Robustness Signal processing Nonlinear distortion Acoustic distortion Content management Data mining Feature extraction audio coding Background noise Fingerprint recognition

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A novel random access approach for MPEG-1 multicast applications

A novel random access approach for MPEG-1 multicast applicat...

引用

International Conferences on Info-tech and Info-net (ICII)

作者： Li xue ming School of Information Engineering of BUPT Beijing China

ISBN: (纸本)0780370104

MPEG-1 is a successful international standard for video and audio coding, and has been widely used in many fields, such as entertainment, education, digital library, video on demand etc. As the MPEG-1 stream has its own semantic structure, the multicast application may destroy its syntax which results in decoding failure. To deal with this problem, a novel random access method is proposed in this paper to repair the corrupted stream structure. This solution consists of two main steps: first, it should extract necessary information from the multicast stream, and then a system header would be constructed using the information, whereafter this header would be inserted to the beginning of the decoder buffer. By this way, the decoder can play the received multicast stream without any problem.

关键词： Decoding Streaming media Video on demand ISO standards IEC standards Software libraries Data mining audio coding Educational programs Timing

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Multi-stream transmission for spectrally and spatially varying interference AM-DAB channels

Multi-stream transmission for spectrally and spatially varyi...

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IEEE International Conference on Communications (ICC)

作者： C.-E.W. Sundberg H. Lou D. Sinha Agere Systems Murray Hill NJ USA Agere Systems NJ

Hybrid in band on channel (IBOC) digital audio broadcasting (DAB) simultaneously with analog amplitude modulation (AM) has been proposed as a hybrid solution to digital audio broadcasting in the AM band. Adding digital transmission in the crowded AM band is a challenging proposition. To achieve FM like audio quality, an audio coder rate of 32-64 kbit/s may be required. One of the currently proposed hybrid IBOC-AM systems is 30 kHz wide. Severe second adjacent interference may occur in certain geographical areas. For coping with such harsh transmission conditions, we present a solution based on embedded/multi-descriptive audio coding with matched multistream transmission in separate frequency bands. With loss of one frequency band, the embedded system blends to a lower audio coder rate with still a better quality than analog AM. The non-embedded system without multi-stream transmission fails catastrophically causing a severe discontinuity in quality while blending directly to analog AM.

关键词： Interference Frequency modulation audio coding Robustness Bit rate Modems Modulation coding Forward error correction Digital audio broadcasting Propagation losses

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An improved error resilience scheme for transmission of MPEG-4 audio over EGPRS

An improved error resilience scheme for transmission of MPEG...

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IEEE Conference on Vehicular Technology (VTC)

作者： Lei Miao Jianhua Lu Jun Gu Department of Electronic Engineering Tsinghua University Beijing China

An improved pre-sorting and reordering algorithm is proposed for enhancing the quality of transmitting real-time MPEG-4 AAC (advanced audio coding) audio over EGPRS (Enhanced General Packet Radio Service). Compared with the standard Huffman codeword reordering (HCR) algorithm, lower implementation complexity and much better audio signal reconstruction quality may be achieved with the proposed scheme.

关键词： Resilience MPEG 4 Standard Bit error rate Psychoacoustic models audio coding Code standards Signal reconstruction Protection Packet radio networks 3G mobile communication

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A fast bit allocation algorithm for MPEG audio encoder

A fast bit allocation algorithm for MPEG audio encoder

引用

International Symposium on Intelligent Multimedia, Video and Speech Processing

作者： Kai-Tat Fung Yui-Lam Chan Wan-Chi Siu Centre for Multimedia Signal Processing Department of Electronic and Information Engineering Hong Kong Polytechnic University Hung Hom Hong Kong China

ISBN: (纸本)9628576623

A fast bit allocation algorithm for the MPEG audio encoder is proposed. The proposed algorithm employs the bit allocation information of the previous audio frame as a reference for allocating the restricted bits to each of the 32 subbands in the current audio frame such that the number of iterations can be significantly reduced. A process of bit reallocation is also suggested to ensure the generation of an identical MPEG bitstream produced by the standard bit allocation algorithm described in the MPEG audio standard. The result shows that the speed-up of the proposed algorithm is remarkable at different encoded bitrates.

关键词： Bit rate Signal processing algorithms Masking threshold audio coding Quantization Psychoacoustic models Signal to noise ratio Filters Ear Noise level

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Efficient representation of spatial audio using perceptual parametrization

Efficient representation of spatial audio using perceptual p...

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： C. Faller F. Baumgarte Media Signal Processing Research Agere Systems Research Murray Hill NJ USA

We introduce a new scheme for simultaneous placement of a number of sources in auditory space. The scheme is based on an assumption about the relevance of localization cues in different critical bands. Given the sum signal of a number of sources, i.e. a monophonic signal, and a set of parameters (side-information) the scheme is capable of generating a binaural signal by spatially placing the sources contained in the monophonic signal. Potential applications for the scheme are multi-talker desktop conferencing and audio coding. Preliminary experimental results suggest that the listener's ability to identify messages in a multi-talker environment significantly improves by enhancing a monophonic signal with the proposed scheme.

关键词： Signal synthesis Signal processing Signal generators Ear Transfer functions audio coding Magnetic heads audio recording Headphones Decoding

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Improved modeling of audio signals by modifying transient locations

Improved modeling of audio signals by modifying transient lo...

引用

IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： R. Vafin R. Heusdens S. van de Par W. Bastiaan Kleijn Department of Speech Music and Hearing KTH Royal Institute of Technology Stockholm Sweden Department of Mediamatics Delft University of Technnology Delft Netherlands Digital Signal Processing Group Philips Research Laboratories Eindhoven Netherlands

We propose a method for obtaining an improved representation of transients in audio signals. The representation is based on a damped sinusoidal model. To improve the representation, transient locations are modified in such a way that a transient can start only at the beginning of a sinusoidal segment. The introduced modifications facilitate a reduction of the number of damped sinusoids needed to model a transient well and eliminate pre-echo artifacts. We verify with a listening test that the modifications do not result in a perceptual difference between the original and modified audio signals.

关键词： Signal processing Testing Signal analysis Speech processing Multiple signal classification Auditory system Laboratories Bit rate Algorithm design and analysis audio coding

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