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IEEE International Symposium on Signal Processing and Information Technology (ISSPIT)

作者： Gaoyong Luo Faculty of Technology Buckinghamshire Chilterns University College Buckinghamshire UK

Wireless systems are often subject to bandwidth or cost constraints which are incompatible with high data rates. The key enabling technology for digital audio wireless products is data compression. For real time wireless transmission, very low encoding and decoding delay has become an essential prerequisite. In live productions, the tolerable total delay time is less than a few milliseconds. Current audio coding schemes like MPEG standards or wavelet techniques can hardly reach such a threshold by using overlapping frames of input signal with psychoacoustic model. This paper presents a two dimensional (2D) spatial-frequency processing based audio coder with ultra low delay for real time wireless applications using non-overlapping short block processing and embedded coding. 2D fast lifting wavelet transform with boundary effects minimized is developed for further exploring the correlation of the audio signal. A modified 2D SPIHT (set partitioning in hierarchical trees) algorithm with more bits used to encode the wavelet coefficients and transmitting fewer bits in the sorting pass, is implemented to reduce the correlation between the coefficients at different decomposition levels and inside each band at scalable bit rates. The experiment shows the proposed coder is efficient and has low complexity with less memory requirements in implementation

关键词： audio coding Delay effects Bandwidth Costs Data compression Encoding Decoding Production MPEG standards Psychoacoustic models

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Unified speech and audio coding scheme for high quality at low bitrates

Unified speech and audio coding scheme for high quality at l...

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

作者： M. Neuendorf P. Gournay M. Multrus J. Lecomte B. Bessette R. Geiger S. Bayer G. Fuchs J. Hilpert N. Rettelbach R. Salami G. Schuller R. Lefebvre B. Grill Fraunhofer IIS Erlangen Germany University of Sherbrooke Sherbrooke Canada Voice Age Corporation Montreal Canada Fraunhofer IDMT Ilmenau Germany

ISBN: (纸本)9781424423538

Traditionally, speech coding and audio coding were separate worlds. Based on different technical approaches and different assumptions about the source signal, neither of the two coding schemes could efficiently represent both speech and music at low bitrates. This paper presents a unified speech and audio codec, which efficiently combines techniques from both worlds. This results in a codec that exhibits consistently high quality for speech, music and mixed audio content. The paper gives an overview of the codec architecture and presents results of formal listening tests comparing this new codec with HE-AAC(v2) and AMR-WB+. This new codec forms the basis of the reference model in the ongoing MPEG standardization activity for unified speech and audio coding.

关键词： Speech coding audio coding Bit rate Speech codecs Psychoacoustic models Predictive models Multiple signal classification Testing Frequency domain analysis Quantization

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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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HIGH-QUALITY audio TRANSFORM coding AT 64-KBPS

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IEEE TRANSACTIONS ON COMMUNICATIONS 1994年第11期42卷 3010-3019页

作者： MAHIEUX, Y PETIT, JP Centre National d''Etudes Des Telecommunications Lannion France

This paper presents a transform coding algorithm devoted to high quality audio coding at a bit rate of 64 kbps per monophonic channel. It enables the transmission of a high quality stereo sound through the basic access (2B channels) of ISDN. Although a complete system including framing, synchronization and error correction has been developed, only the bit rate compression algorithm is described here. A detailed analysis of the signal processing techniques such as the time/frequency transformation, the pre-echo reduction by adaptive filtering, the fast algorithm computations, etc., is provided. The use of psychoacoustical properties is also precisely reported. Finally, some subjective evaluation results and one real time implementation of the coder using the ATT DSP32C digital signal processor are presented.

关键词： Transform coding Signal processing algorithms Bit rate audio coding ISDN Frequency synchronization Error correction Compression algorithms Signal analysis Algorithm design and analysis

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Trellis-based optimization of MPEG-4 advanced audio coding

Trellis-based optimization of MPEG-4 advanced audio coding

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IEEE Workshop on Speech coding Proceedings

作者： A. Aggarwal S.L. Regunathan K. Rose Department of Electrical and Computer Engineering University of California Santa Barbara CA USA

We outline a method to perform efficient low rate quantization for MPEG-4 advanced audio coding (AAC). The AAC bit stream consists of indices for quantized spectral coefficients as well as side information about quantizer step sizes and Huffman codebooks. The MPEG-4 Verification Model does not explicitly account for side information bits in its optimization and suffers from poor compression efficiency at low bit rates. We reformulate the encoding problem as one of optimal parameter selection, where the side information bits are taken into account, so as to minimize the noise to mask ratio for the given target bit rate. The optimal solution is determined by a dynamic programming procedure that efficiently searches through a trellis. This trellis-based optimization greatly improves the low bit rate performance of AAC and, consequently, the performance of a multi-layer AAC system. The resulting bit stream is standard-compatible, and additional complexity due to the proposed optimization is only incurred at the encoder.

关键词： MPEG 4 Standard audio coding Bit rate Encoding Quantization Streaming media Signal to noise ratio Dynamic programming audio compression Virtual manufacturing

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A Survey on Multimedia Communicating Technology Based on Spatial audio coding

A Survey on Multimedia Communicating Technology Based on Spa...

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International Conference on Advanced Information Networking and Applications (AINA)

作者： Naixue Xiong Shuixian Chen Jing He Yanxiang He Athanasios T. Vasilakos Jong Hyuk Park Yan Yang School of Computer Science & Technology Wuhan University of Science and Engineering Wuhan China National Engineering Research Center for Multimedia Software Wuhan University of China Wuhan China Department of Computer Science Georgia State University Atlanta USA Computer School Wuhan University of China Wuhan Hubei China Dept of Computer and Teleco. Engineering University of Western Macedonia Greece Department of Computer Science and Engineering Seoul National University of Technology South Korea Department of Computer Science Western Illinois University IL USA

Spatial audio coding (SAC) is an emerging technology with a distinguishing feature of delivering good even excellent audio quality at monotonic or stereo bitrate of conventional perceptual transform coders. By a systematic exploitation of spatial hearing, Binaural Cue coding illustrates the power and potentials of SAC in the future for intelligent multimedia services. MPEG Surround, receiving cumulative efforts from industry and academy, strives to build a SAC system with great versatility and high quality. The initial test results of MPEG Surround show its performance advantage over conventional state-of-the-art coders in a wide range of coding configurations.

关键词： Multimedia communication audio coding Bit rate Auditory system Computer science USA Councils Computer networks Testing Transform coding Application software

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An embedded speech and audio coding method based on bit-plane coding and SQVH

An embedded speech and audio coding method based on bit-plan...

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IEEE International Symposium on Signal Processing and Information Technology (ISSPIT)

作者： Mao-shen Jia Chang-chun Bao Xin Liu Speech and Audio Signal Processing Laboratory School of Electronic Information and Control Engineering Beijing University of Technology Beijing China

A multi-layer embedded speech and audio coding algorithm based on bit-plane coding and Scalar Quantized Vector Huffman coding (SQVH) is proposed in this paper. In this codec the signal sampled at 32 kHz can be coded in terms of scalable bit rates. The core codec is International Telecommunication Union Telecommunication Standardization Sector (ITU-T) G.729.1 which can process signal with 7 kHz bandwidth. Besides there are five extra bit-rates added and the bandwidth is extended to 14 kHz. The additional bit-rates include 36, 40, 48, 56, and 64 kb/s. Some new methods used in the additional layers are proposed in this paper. The objective and subjective listening tests show that this codec has good performance compared with reference codec.

关键词： Speech coding audio coding Codecs Speech processing Signal processing Bandwidth Wideband Signal processing algorithms Standardization Mobile communication

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Predictive audio coding using rate-distortion-optimal pre- and post-filtering

Predictive audio coding using rate-distortion-optimal pre- a...

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics

作者： Obada Alhaj Moussa Minyue Li W. Bastiaan Kleijn Electrical Engineering Royal Institute of Technology KTH Stockholm Sweden School of Engineering and Computer Science Victoria University of Wellington New Zealand

A natural approach to audio coding is to use a rate-distortion optimal design combined with a perceptual model. While this approach is common in transform coding, existing predictive-coding based audio coders are generally not optimal and they benefit from heuristically motivated post-filtering. As delay requirements often force the use of predictive coding, we consider audio coding with a pre- and post-filtered predictive structure that was recently proven to be asymptotically optimal in the rate-distortion sense [1]. We show that this audio coding is efficient in achieving the state-of-the-art performance. We also show that the pre-filter plays a relatively minor role. This leads to an analytic approach for optimizing the post-filter and the predictor at each rate, eliminating the need for manual re-tuning whenever a different rate is called for. In a subjective test, the theoretically optimized post-filter provided a better performance than a conventional post-filter.

关键词： Quantization Noise Speech Resource description framework audio coding Rate-distortion

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Scalable audio coding using the Oversampled Discrete Wavelet Transform and Temporal Masking

Scalable Audio Coding using the Oversampled Discrete Wavelet...

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International Conference on Information, Communications and Signal Processing

作者： F. Sinaga T.S. Gunawan E. Ambikairajah School of Electrical Engineering and Telecommunications University of New South Wales Sydney NSW Australia

In this paper, we propose a low-complexity scalable audio coding algorithm using an over sampled discrete wavelet transform (ODWT). The aliased free coefficients obtained are then used to accurately calculate the forward temporal masking threshold. The algorithm based on the bandwidth scalability, uses bits per sub-band and bits per frame, as outlined in this paper. Subjective tests conforming to ITU-R BS.1116 for audio signals show that the audio signals coded at 32 and 64 kbits/s using an ODWT with temporal masking offered a better audio quality compared to the critically sampled DWT

关键词： Discrete wavelet transforms audio coding Masking threshold Wavelet coefficients Bit rate Bandwidth Decoding Personal communication networks Scalability audio compression

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A psychoacoustic model for audio coding based on a cochlear filter bank

A psychoacoustic model for audio coding based on a cochlear ...

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IEEE Workshop on Applications of Signal Processing to audio and Acoustics

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

Perceptual audio coders use an estimated masked threshold for the determination of the maximum permissible just-inaudible noise level introduced by quantization. This estimate is derived from a psychoacoustic model mimicking the psychoacoustics of masking. Current applications use a uniform spectral decomposition as first stage of that model to approximate the frequency selectivity of the human auditory system. The availability of efficient implementations led to a virtually exclusive use of uniform decompositions in audio coding. However, the equal filter properties of the uniform sub-bands are not in line with the nonuniform auditory filters. This paper presents a psychoacoustic model based on an efficient nonuniform cochlear filter bank with a simplified less complex post-processing for estimating the masked threshold. Application results in audio coding show a significantly better performance in terms of bit rate and/or quality of the new model in comparison with other state-of-the-art models with a uniform spectral decomposition.

关键词： Psychoacoustic models audio coding Filter bank Noise level Quantization Psychology Frequency Humans Auditory system Bit rate

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