检索结果-内蒙古大学图书馆

Biorthogonal and nonuniform lapped transforms for transform coding with reduced blocking and ringing artifacts

IEEE TRANSACTIONS ON SIGNAL PROCESSING 1998年第4期46卷 1043-1053页

作者： Malvar, HS PictureTel Corp Andover MA 01810 USA

New lapped transforms are introduced, The lapped biorthogonal transform (LET) and hierarchical lapped biorthogonal transform (HLBT) are appropriate for image coding, and the modulated lapped biorthogonal transform (MLBT) and nonuniform modulated lapped biorthogonal transform (NMLBT) are appropriate for audio coding, The HLBT has a significantly lower computational complexity than the lapped orthogonal transform (LOT), essentially no blocking artifacts, and fewer ringing artifacts than the commonly used discrete cosine transform (DCT), The LET and HLBT have transform coding gains that are typically between 0.5 and 1.2 dB higher than that of the DCT, Image coding examples using JPEG and embedded zerotree coders demonstrate the better performance of the LET and HLBT. The NMLBT has fewer ringing artifacts and better reproduction of transient sounds than the MLT, as shown in audio coding examples, Fast algorithms for both the HLBT and the NMLBT are presented.

关键词： audio coding image coding lapped transforms speech coding transforms

来源：评论

学校读者我要写书评

暂无评论

coding of natural audio in MPEG-4

Coding of natural audio in MPEG-4

引用

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

作者： S.R. Quackenbush AT and T Research Laboratories Florham Park NJ USA

MPEG-4 standardizes natural audio coding at bit rates ranging from 2 kbit/s, suitable for intelligible speech coding, to 64 kbit/s per channel, suitable for high-quality audio coding. Within this range, three categories of coding are defined: parametric coding, code excited linear predictive coding (CELP) and time/frequency (T/F) coding. The unique contribution of MPEG-4 audio is that not only does it scale across a wide range of bit rates, but it also scales across a broad set of other parameters, such as sampling rate, bandwidth, voice pitch and complexity. This paper presents an overview of the MPEG-4 natural audio coding framework and each of its component coding techniques.

关键词： MPEG 4 Standard Bit rate audio coding Decoding Speech coding Scalability Sampling methods Bandwidth Frequency Signal synthesis

来源：评论

学校读者我要写书评

暂无评论

An OFDM all digital in-band-on-channel (IBOC) AM and FM radio solution using the PAC encoder

引用

IEEE TRANSACTIONS ON BROADCASTING 1998年第1期44卷 22-27页

作者： Cupo, RL Sarraf, M Shariat, M Zarrabizadeh, M AT&T Bell Labs Lucent Technol Holmdel NJ 07733 USA

Recent advances in digital communications and compression algorithms have made more efficient and more robust transmission schemes possible. Radio broadcast systems have not fully utilized these advances to their benefit. All digital robust radio broadcast systems for the AM and the FM band are proposed in this paper. The proposed systems are based on Orthogonal Frequency Division Multiplexing (OFDM) technology in conjunction with PAC for both the AM and the FM bands. The Perceptual audio Coder (PEG) developed by Bell Laboratories compresses audio signals very efficiently with CD-like quality at 96 kbps and stereo FM-like quality at 48 kbps. These are rates achievable with 200 kHz and 30 kHz bandwidths available per FM and AM station respectively, No new spectrum is required since the digital signals are transmitted within the current allocated FCC masks. Tn an FM channel, a wide-band data subchannel can be provided in addition to the 96 kbps error protected PAC audio information.

关键词： Hi-Fi equipment amplitude modulation audio coding audio signals data compression digital audio broadcasting digital signals frequency division multiplexing frequency modulation telecommunication channels 200 kHz 30 kHz 48 kbit/s 96 kbit/s AM band AM radi Orthogonal frequency division multiplexing audio coding amplitude modulation digital audio broadcasting Frequency division multiplexing Data compression Hi-Fi equipment FM band perturbed angular correlation digital signals audio signals

来源：评论

学校读者我要写书评

暂无评论

Efficient audio coding using perfect reconstruction noncausal IIR filter banks

引用

IEEE TRANSACTIONS ON SPEECH AND audio PROCESSING 1996年第2期4卷 115-123页

作者： Creusere, CD Mitra, SK Weapons Division Naval Air Warfare Center China Lake CA USA Department of Electrical and Computer Engineering University of California Santa Barbara CA USA

In this paper, we consider the problem of implementing perfect reconstruction modulated polyphase filter banks having infinite impulse response (IIR) subfilters in systems that must process infinite length 1-D inputs. This task is complicated by the fact that the synthesis bank needed to ensure perfect reconstruction in such a system generally has noncausal or reverse-time filter sections. To overcome this limitation, we develop a number of unique buffering schemes that make possible the implementation of such reverse-time filters with infinite length signals. For the important special case when the polyphase subfilters are allpass, we also present a method that allows for very efficient implementation of tree-structured filter banks and wavelet transforms. We then describe a wideband coding algorithm based largely on work done by Lokhoff and Johnston. Using this algorithm, we compare the performance of IIR filter banks with that of more conventional pseudo-QMF and tree-structured QMF banks. The results of these comparisons indicate that one of our filter banks-a tree-structured allpass bank-achieves slightly better performance than equivalent FIR banks in terms of rate and distortion (both objectively and subjectively measured) while requiring lower computational complexity.

关键词： audio coding Filter bank IIR filters Finite impulse response filter Signal synthesis Wavelet transforms Wideband Rate distortion theory Distortion measurement Computational complexity

来源：评论

学校读者我要写书评

暂无评论

Structured audio: Creation, transmission, and rendering of parametric sound representations

引用

PROCEEDINGS OF THE IEEE 1998年第5期86卷 922-940页

作者： Vercoe, BL Gardner, WG Scheirer, ED MIT Media Lab Machine Listening Grp Cambridge MA 02139 USA Wave Arts Inc Arlington VA 02174 USA

Structured audio representations are semantic and symbolic descriptions that are useful for ultralow-bit-rate transmission, flexible synthesis, and perceptually based manipulation and retrieval of sound. We present an overview of techniques for transmitting and synthesizing sound represented in structured format, and for creating structured representations from audio waveforms. We discuss applications for structured audio in virtual environments, music synthesis, gaming, content-based retrieval, interactive broadcast, and other multimedia contexts.

关键词： audio coding audio systems digital audio broadcasting multimedia computing multimedia data bases music

来源：评论

学校读者我要写书评

暂无评论

Regressive implementations for the forward and inverse MDCT in MPEG audio coding

引用

IEEE SIGNAL PROCESSING LETTERS 1996年第4期3卷 116-118页

作者： Chiang, HC Liu, JC Department of Electrical Engineering Tatung Institute of Technology Taipei Taiwan

Modified discrete cosine transform (MDCT) and inverse modified discrete cosine transform (IMDCT) are two of the most compute-intensive operations in layer III of the MPEG audio coding standard, In this letter, we first derive two sinusoidal recursive formulas for the transform kernels of the MDCT and IMDCT. Then we demonstrate that general length of MDCT and IMDCT can be efficiently implemented by using the regressive structure derived from the sinusoidal recursive formulas, The proposed regular structure is particularly suitable for parallel VLSI realization, Besides, it also provides an efficient on-line computation scheme and is especially suitable for the computation of variable length MDCT and IMDCT when involving dynamical window switching in MPEG audio coding standard.

关键词： audio coding Very large scale integration Hardware

来源：评论

学校读者我要写书评

暂无评论

Real-time implementation of the MPEG-2 audio codec on a DSP

引用

IEEE TRANSACTIONS ON CONSUMER ELECTRONICS 1998年第3期44卷 866-871页

作者： Chen, J Tai, HM Univ Tulsa Dept Elect Engn Tulsa OK 74104 USA

This piper presents the implementation of MPEG-2 layer 1 audio coding and decoding system on a general purpose fixed-point DSP chip. A modified coding algorithm th:lt alleviates the edge effect is described. This scheme encodes data in frames adjacent to the current frame so that data is closely correlated between frames. Results from the DSP realtime implementation are examined. Bit-rates from 74 to 138 kbps for the signals tested are attainable with transparent quality signal reconstruction.

关键词： Codecs Digital signal processing Decoding Digital signal processing chips Quantization Sampling methods Frequency audio coding Real time systems Bit rate

来源：评论

学校读者我要写书评

暂无评论

Fast time-frequency transform algorithms and their applications to real-time software implementation of AC-3 audio codec

引用

IEEE TRANSACTIONS ON CONSUMER ELECTRONICS 1998年第2期44卷 413-423页

作者： Chen, YC Tsai, CW Wu, JL Natl Taiwan Univ Dept Comp Sci & Informat Engn Taipei 10764 Taiwan

AC-3 audio coding technology is a kind of Perceptual audio Coder (PAC) developed by the Dolby Company. Up to 5 full-bandwidth channels and one subwoofer channel (cutoff at 120Hz) are available in AC-3 to provide multi-channel, low bit rate, and high perceptual quality of audio. This explains why AC-3 has become the audio standard of many international standards. In this paper, we focus on the real-time software implementation issues of AC-3, Two fast algorithms of the time-frequency transform, one for memory economization and the other is for time domain subsampling, are presented. Meanwhile, we will state the current performance status of our AC-3 decoder.

关键词： Time frequency analysis Software algorithms Application software Psychology Bit rate Psychoacoustic models Signal processing algorithms audio coding Decoding Multimedia systems

来源：评论

学校读者我要写书评

暂无评论

Multi-domain speech compression based on wavelet packet transform

引用

ELECTRONICS LETTERS 1998年第2期34卷 154-155页

作者： Wu, XD Li, YM Chen, HY Tsing Hua Univ Inst Microelect Beijing 100084 Peoples R China

The authors present a multi-domain speech compression method based on a wavelet packet transform. The signals are compressed in domains with different time-frequency resolutions according to their energy distribution in these domains. It is shown that this method is simple to implement and is effective at compressing speech and audio signals, even at bit rates as low as 2kbit/s.

关键词： wavelet transforms multi-domain speech compression audio signals audio coding Integral transforms data compression energy distribution time-frequency resolutions Codes speech coding Speech and audio signal processing Information theory Signal processing and detection 2 kbit/s wavelet packet transform

来源：评论

学校读者我要写书评

暂无评论

Efficient bit allocation algorithm for ISO/MPEG audio encoder

引用

ELECTRONICS LETTERS 1998年第8期34卷 721-722页

作者： Teh, DH Koh, SN Tan, AP Nanyang Technol Univ Sch Elect & Elect Engn Singapore 2263 Singapore Panason Singapore Labs Pte Ltd Singapore Singapore

An efficient bit allocation algorithm for the ISO/MPEG audio layer 2 encoder is proposed. The resulting encoder is able to produce an ISO/MPEG compliant bit stream which is almost identical to the bit stream produced by the reference ISO/MPEG encoder suggested in IS 11172-3. More importantly, the number of computational steps is greatly reduced as compared to the method recommended by the ISO/MPEG Committee because the efficient bit allocation algorithm significantly reduces the number of iterations required. It was found that the efficient bit allocation algorithm works best when the bit rate demanded by the psychoacoustic model in order to keep the quantisation noise below the masking threshold is almost equal to the operational bit rate.

关键词： ISO/MPEG audio encoder telecommunication standards operational bit rate audio coding data compression ISO standards bit allocation algorithm masking threshold quantisation noise ISO/MPEG compliant bit stream Codes quantisation (signal) computational steps reduction audio equipment and systems Signal processing and detection 96 kbit/s psychoacoustic model

来源：评论

学校读者我要写书评

暂无评论

建议与咨询留下您的常用邮箱和电话号码，以便我们向您反馈解决方案和替代方法

时间限定

文献类型

馆藏选择

核心期刊

语言

文献类型

帮助

文字说明：

检索规则说明：

检索范例：

分类表

所选分类

限定检索结果

文献类型

馆藏范围

日期分布

学科分类号

主题

机构

作者

语言

请选择保存的检索档案：

请选择收藏分类：

通借通还

建议与咨询 留下您的常用邮箱和电话号码，以便我们向您反馈解决方案和替代方法

时间限定

文献类型

馆藏选择

核心期刊

语言

文献类型

帮助

文字说明：

检索规则说明：

检索范例：

分类表

所选分类

限定检索结果

文献类型

馆藏范围

日期分布

学科分类号

主题

机构

作者

语言

请选择保存的检索档案： 新增检索档案 确定 取消

请选择收藏分类： 新增自定义分类 确定 取消

通借通还

建议与咨询留下您的常用邮箱和电话号码，以便我们向您反馈解决方案和替代方法

请选择保存的检索档案：

请选择收藏分类：