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On the applications of multimedia processing to communications

PROCEEDINGS OF THE IEEE 1998年第5期86卷 755-824页

作者： Cox, RV Haskell, BG Lecun, Y Shahraray, B Rabiner, L AT&T Bell Labs Speech & Image Proc Serv Res Lab Florham Pk NJ 07932 USA

The challenge of multimedia processing is to provide services that seamlessly integrate text sound, image, and video information and to do it in a way that preserves the ease of use and interactivity of conventional plain old telephone service (POTS) telephony, irrelevant of the bandwidth or means of access of the connection to the sen,ice. To achieve this goal, there are a number of technological problems that must be considered, including: compression and coding of multimedia signals, including algorithmic issues, standards issues, and transmission issues;synthesis and recognition of multimedia signals, including speech, images, handwriting, and text;organization, storage, aid retrieval of multimedia signals, including the appropriate method and speed of delivery (e.g., streaming versus full downloading), resolution (including layering or embedded versions of the signal), and quality of service, i.e., perceived quality of the resulting signal;access methods to the multimedia signal (i.e., matching the user to the machine), including spoken natural language interfaces, agent interfaces, and media conversion tools;searching (i.e., based on machine intelligence) by text, speech, and image queries;browsing (i.e., based on human intelligence) by accessing the text, by voice, or by indexed images. In each of these areas, a great deal of progress has been made in the past few years, driven in parr by the relentless growth in multimedia personal computers and in part by the promise of broad-band access from the home and from wireless connections. Standards have also played a key role in driving new multimedia services, both on the POTS network and on the Internet. It is the purpose of this paper to review the status of the technology in each of the areas listed above and to illustrate current capabilities by describing several multimedia applications that have been implemented at AT&T Labs over the past several years.

关键词： AAC access agents audio coding cable modems communications networks content-based video sampling document compression fax coding H.261 HDTV image coding image processing JBIG JPEG media conversion MPEG multimedia multimedia browsing multimedia indexing multimedia searching optical character recognition PAC packet networks perceptual coding POTS telephony quality of service speech coding speech compression speech processing speech recognition speech synthesis spoken language interface spoken language understanding standards streaming teleconferencing video coding video telephony

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audio transfer system on phs using error-protected stereo twin VQ

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IEEE TRANSACTIONS ON CONSUMER ELECTRONICS 1998年第3期44卷 1032-1038页

作者： Ikeda, K Mori, T Moriya, T Iwakami, N Kaneko, T NTT Human Interface Labs Tokyo Japan

We propose an efficient stereo audio coding scheme with error protection based on transformed vector quantization. This scheme includes a new stereo coding method, called adaptive inter-channel prediction stereo, that achieves high-efficiency stereo coding at a low coding bit rate. This scheme also includes an unequal error protection (UEP) method that achieves high-efficiency error protection against radio transmission error. We also report a real-time audio transfer system on the Japanese Personal Handyphone System (PHS) using the proposed coding scheme, which enables a high-quality audio transfer service over mobile channels.

关键词： Automatic repeat request audio coding Bit rate Error correction codes Vector quantization Real time systems Wireless communication Transform coding Telephony Protection

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Cycle saving hardware for real-time audio processing

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ELECTRONICS LETTERS 1998年第9期34卷 847-848页

作者： Park, SW Yoo, SK Jeong, NH Kim, JS Ko, WS Lee, KS Youn, DH Yonsei Univ Dept Elect Engn ASSP Lab Sudaemun Ku Seoul 120749 South Korea

The authors present a design for cycle saving hardware components that enable real-time operation of audio coder-decoders. To determine the key components of audio coding algorithms, the MPEG2 audio algorithm is analysed, which is an international standard and uses a mixture of several coding strategies. Through the analysis, some components are selected that significantly contribute to the total number of cycles, and corresponding hardware accelerators that complete assigned tasks in a single cycle are designed. By incorporating these accelerators, a simple processor can reduce the number of cycles for assigned tasks by up to 72.2%.

关键词： Digital signal processing chips Microprocessors and microcomputers Digital signal processing digital signal processing chips DSP cores hardware accelerators Codecs, coders and decoders data compression audio coding algorithms MPEG2 audio algorithm Codes audio equipment and systems real-time systems audio coder-decoders audio coding real-time audio processing telecommunication computing codecs cycle saving hardware Signal processing and detection Communications computing

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audio coding using variable-depth multistage quantizers

Audio coding using variable-depth multistage quantizers

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Data Compression Conference (DCC)

作者： F. Kossentini M.W. Macon M.J.T. Smith Department of EE University of British Columbia Vancouver BC Canada School of ECE Georgia Tech Atlanta GA USA

Digital coding of high-fidelity audio signals has become a key technology in the development of cost-effective multimedia systems. Most audio coding algorithms rely on (i) removal of statistical redundancies in the signal and (ii) exploitation of masking properties of the human auditory system to hide distortions in the coded signal. Transform and subband coders provide a convenient framework for time-frequency domain signal analysis and coding based on these two principles. We present a subband coding algorithm for high-quality coding of 48 kHz sampled audio signals that is based on perceptual transformation, variable-depth multistage uniform quantization, and complexity-constrained higher-order entropy coding. The construction does not require any training and avoids completely the need to send side information.

关键词： audio coding Quantization Multi-stage noise shaping Humans Multimedia systems Auditory system Distortion Time frequency analysis Signal analysis Entropy coding

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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...

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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 with a dynamic wavelet packet decomposition based on frequency-varying modulated lapped transforms

Audio coding with a dynamic wavelet packet decomposition bas...

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

作者： M. Purat P. Noll Institute for Telecommunications Technical University Berlin Berlin Germany

Optimum time-frequency decompositions are very useful in audio coding applications, because the signal energy can be maximally concentrated even for the wide variety of audio signal characteristics. Moreover, this signal representation is particularly well suited for a perceptual weighting of the quantization noise. The well known tree structure of cascaded 2-channel filterbanks allows a very flexible optimization, leading to a signal adaptive, dynamic wavelet packet decomposition. A major drawback of this technique are strong spectral side lobes which produce clearly audible aliasing in perceptual coders. We present a new dynamic wavelet packet decomposition, based on modulated lapped transforms, which allows the same flexibility while avoiding the disadvantage mentioned above. We propose a scheme for low bit rate audio coding that efficiently exploits the high energy concentration. This new codec yields excellent audio quality at about 55 kb/s for monophonic signals.

关键词： audio coding Wavelet packets Time frequency analysis Signal representations Quantization Acoustic noise Tree data structures Adaptive filters Wavelet transforms Bit rate

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An adaptive tree search algorithm with application to multiresolution based perceptive audio coding

An adaptive tree search algorithm with application to multir...

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IEEE-SP International Symposium on Time-Frequency and Time-Scale Analysis

作者： N.G. Prelcic A.S. Pena Depart ament o de Tecnoloxías das Comunicacións Universidade de Vigo Vigo Spain

Search algorithms for selecting signal decompositions based on the minimization of a cost functional have been proposed in the literature, in addition to several additive and non-additive information costs. We introduce a new cost function and a search algorithm built from a perceptual criterion. Their efficiency is demonstrated with results showing higher quality compressed audio signals than preliminary approaches at similar bit-rates.

关键词： audio coding Cost function Frequency Bandwidth Signal resolution Wavelet packets Distortion Signal analysis Auditory system Minimization methods

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An iterated rational filter bank for audio coding

An iterated rational filter bank for audio coding

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IEEE-SP International Symposium on Time-Frequency and Time-Scale Analysis

作者： T. Blu France-Télécom-CNET Issy-les-Moulineaux France

This paper proposes a regular third-of-an-octave filter bank for high fidelity audio coding. The originality here is twofold: first, the filter bank is an iterated orthonormal rational filter bank for which the generating filters have been designed so that its outputs closely approximate a wavelet transform. This is different from the known coding algorithms which all use an integer filter bank, and most often a uniform one; second, the masking procedure itself is modelized with the help of a wavelet transform unlike the classical procedure in which a short time spectrum is computed and which gives rise to unwanted preecho effects. The masking procedure is then made equivalent to a quantization procedure. A simple non-optimized algorithm has been worked out in order to show the benefits of such a structure, especially in terms of preecho (which is perceptually inaudible), and the disadvantages, especially as far as delay is concerned.

关键词： Filter bank audio coding Discrete wavelet transforms Finite impulse response filter Wavelet transforms Algorithm design and analysis Ear Frequency Wavelet analysis Discrete transforms

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Optimal wavelet packets for low-delay audio coding

Optimal wavelet packets for low-delay audio coding

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

作者： P. Philippe F. Moreau de Saint-Martin M. Lever J. Soumagne CCETT France SUPELEC Metz France

We address the issue of choosing an optimal wavelet packets decomposition for low-delay audio bit-rate reduction. The decomposition tree and the choice of the wavelet filters for each 2-band cell of the tree are optimized. For a given filterbank we measure the entropy of the symbols resulting from a quantization stage ensuring transparent compression. This measure is the criterion to be minimized under a delay constraint. The optimization is carried out with simulated annealing techniques among a large family of filters. We show that there is no loss choosing the wavelet filters among a family of filters that have been optimized with respect to coding gain for AR(1) models. This provides efficient wavelet packets for low bit-rate low delay audio compression.

关键词： Wavelet packets audio coding Filter bank Entropy Propagation delay Quantization Simulated annealing audio compression Degradation Noise reduction

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Low bit rate high quality audio coding with combined harmonic and wavelet representations

Low bit rate high quality audio coding with combined harmoni...

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

作者： K.N. Hamdy M. Ali A.H. Tewfi Department of Electrical Engineering University of Minnesota Minneapolis MN USA University of Minnesota Minneapolis MN US

We describe a novel high quality audio coding method using adaptive signal representation, based on sinusoidal and wavelet analysis of signals. First, we perform a harmonic analysis of the signal to remove strong periodic structures or tones from the signal. Then we carry out wavelet analysis that are useful in tracking the transients of the signal. These transients are then removed from the wavelet coefficients. The remaining coefficients have broadband noise-like structure. Since this method separates out tones (sinusoids), transients, and broadband noise, we may use tonal, noise, and temporal masking information to individually encode the tones and the wavelet coefficients. Our experiments suggest that this method yields a nominal bit rate of 1 bit/sample for high quality audio compression.

关键词： Bit rate audio coding Wavelet analysis Signal analysis Wavelet coefficients Signal representations Harmonic analysis Periodic structures Transient analysis audio compression

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