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检索条件"主题词=Audio coding"

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MPEG-2 AAC decoder for a 32-bit MCU

MPEG-2 AAC decoder for a 32-bit MCU

引用

IEEE International Conference on Consumer Electronics (ICCE)

作者： T. Sakamoto M. Taruki T. Hase System LSI Development Center Mitsubishi Electric Corporation Limited Hyogo Japan

New implementation techniques for MPEG-2 AAC software by using a 32-bit general-purpose MCU and DRAM are proposed. These make it possible to realize real-time decoding on an evaluation board.

关键词： Decoding Random access memory Acceleration audio coding Gain control Large scale integration MPEG 4 Standard Digital audio broadcasting Digital signal processing Noise figure

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Overview of MPEG-4 audio and its applications in mobile communications

Overview of MPEG-4 audio and its applications in mobile comm...

引用

International Conference on Signal Processing Proceedings (ICSP)

作者： J. Herre B. Grill Audio Department Fraunhofer Institute of Integrated Circuits and Systems Erlangen Germany

MPEG-4 is a member of the well-known family of ISO/MPEG coding standards, and provides an integrated set of audio coders with attractive novel capabilities, including bitrate and bandwidth scalability. The paper gives an introduction into the underlying design concepts of MPEG-4 and provide an overview of MPEG-4 audio coding technology and its features. In particular, it discusses the technical provisions which make MPEG-4 audio very well suited for applications in the domain of mobile communications.

关键词： MPEG 4 Standard Mobile communication audio coding Bit rate Channel capacity ISO standards Code standards Standards development Application specific integrated circuits Bandwidth

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Using MPEG-4 audio for DRM digital narrowband broadcasting

Using MPEG-4 audio for DRM digital narrowband broadcasting

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： M. Dietz T. Mlasko Fraunhofer Institute for Integrated Circuits Erlangen Germany Robert Bosch GmbH Hildesheim Germany

DRM (Digital Radio Mondiale) is an international consortium, working on the definition of a world-wide standard for digital narrowband broadcasting in short- and medium-wave bands. One major requirement for a digital system in these bands is the bandwidth compatibility with existing analogue services. This means that the system needs to operate with a channel bandwidth of 9 or 10 kHz. Therefore DRM has a need for highly efficient audio and speech coding. A further requirement is error robustness. Due to the fading characteristics of the channel, signal errors may occur in spite of digital modulation and channel coding. Last but not least DRM wants to use an international standard to make sure that widespread, well-tested and well-supported algorithms are being used. Following these requirements, DRM has chosen parts of MPEG-4 audio for the DRM system. Based on listening tests performed in cooperation with the NADIB project, the European predecessor of DRM, MPEG has proposed audio and speech coding algorithms suitable for use within DRM. Since then, more work on these algorithms has been done within Version 2 of MPEG-4. MPEG-4 offers highly efficient audio coding by the AAC algorithm as well as state-of-the-art CELP speech coding. Both algorithms are very flexible in terms of codec configuration and bit rate. In addition to that, they support special error robust modes for use in error prone channels. DRM has chosen MPEG-4 AAC for audio coding. The MPEG-4 CELP is a very good candidate for the speech coding part of the system. The DRM specification will be finalized in 2000.

关键词： MPEG 4 Standard Narrowband Digital audio broadcasting Speech coding Radio broadcasting Bandwidth Robustness audio coding Digital Radio Mondiale Digital systems

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Exploiting time and frequency masking in consistent sinusoidal analysis-synthesis

Exploiting time and frequency masking in consistent sinusoid...

引用

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

作者： R. Vafin S.V. Andersen W.B. Kleijn Department of Speech Music and Hearing KTH Royal Institute of Technology Stockholm Sweden

In this paper, we elaborate on the issue of analysis-synthesis consistency in sinusoidal coding. Our analysis is based on windowed sinusoids, and uses the same amplitude-complementary window as is used in the overlap-add synthesis. Reconstructions of the neighboring segments are taken into account when forming a particular analysis segment. Sinusoidal estimation is based on a perceptual criterion. In our new procedure, when analyzing the current segment we take advantage of the forward masking effect due to estimated sinusoids in the previous segments (possibly overlapping with the current segment). Experimental results verify that the number of sinusoids can be reduced significantly with our time masking model, without introducing perceptual artifacts in the reconstructed signal.

关键词： Frequency Psychoacoustic models Speech analysis Bit rate Speech coding Speech synthesis audio coding Performance analysis Auditory system Transform coding

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Theory and design of a class of cosine-modulated non-uniform filter banks

Theory and design of a class of cosine-modulated non-uniform...

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

作者： S.C. Chan X.M. Xie T.I. Yuk Department of Electrical and Electronic Engineering University of Hong Kong Hong Kong China

In this paper, the theory and design of a class of PR cosine-modulated nonuniform filter bank is proposed. It is based on a structure previously proposed by Cox (1986), where the outputs of a uniform filter bank are combined or merged by means of the synthesis section of another filter bank with smaller channel number. Simplifications are imposed on this structure so that the design procedure can be considerably simplified. Due to the use of cosine modulated filter banks as the original and recombination filter banks, excellent filter quality and low design and implementation complexities can be achieved. Problems with these merging techniques such as spectrum inversion, equivalent filter representations and protrusion cancellation are also addressed. As the merging is performed after the decimation, the arithmetic complexity is lower than other conventional approaches. Design examples show that PR nonuniform filter banks with high stopband attenuation and low design and implementation complexities can be obtained by the proposed method.

关键词： Channel bank filters Filter bank Merging Sampling methods Delay Design engineering Arithmetic Attenuation audio coding Adaptive filters

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A 6Kbps to 85Kbps scalable audio coder

A 6Kbps to 85Kbps scalable audio coder

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

作者： T.S. Verma T.H.Y. Meng Laboratory of Acoustics and Audio Signal Processing Helsinki University of Technology Finland Department of Electrical Engineering University of Stanford USA

Scalable audio coding is important in network environments, such as the Internet, where bandwidth is not guaranteed, packet loss is common, and client connection data rates are heterogeneous. Signal models provide a general framework for attacking a wide range of challenges in the unicast delivery of real-time audio over packet switched networks. The specific signal model in this work generates a parametric representation for general wide-band audio signals. The model consists of three complementary components: sines, transients, and noise. Because the human hearing system ultimately judges the validity of a model for audio signals, psychoacoustic principles are explicitly considered in the three part model. Once analyzed, the parameters are quantized, compressed and packed into a single 85Kbps bit-stream. From this bit-stream, bit-streams at several bit-rates between 6Kbps and 85Kbps may be readily extracted. The audio coder offers a wide range of scalability while the audio quality of the coding scheme gracefully degrades from perceptually lossless to low-quality.

关键词： Psychoacoustic models audio coding IP networks Bandwidth Unicast Packet switching Signal generators Wideband Humans Auditory system

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VLSI implementation of portable MPEG-4 audio decoder

VLSI implementation of portable MPEG-4 audio decoder

引用

Annual IEEE International ASIC/SOC Conference

作者： S. Hashimoto A. Niwa H. Okuhata T. Onoye I. Shirakawa Department of Information Systems Engineering Osaka University Suita Osaka Japan Synthesis Corporation Mino Osaka Japan Department of Communications and Computer Engineering Kyoto University Kyoto Japan

VLSI implementation of an MPEG-4 audio decoder is described, which is dedicated to portable audio appliances. MPEG-4 audio surpasses conventional audio codec schemes in terms of sound quality and compression rate. Based on the results of sound quality evaluation, a low-power architecture is devised by means of frame-level pipeline and optimization of functional datapath. The proposed MPEG-4 audio decoder has been implemented with the use of 0.25 /spl mu/m CMOS library, which integrates 30 k gates and dissipates 4.54 mW at 2.5 V supply.

关键词： Very large scale integration MPEG 4 Standard Decoding Bit rate Signal to noise ratio Filter bank audio coding Bandwidth Information systems Systems engineering and theory

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Perceptual phase redundancy in speech

Perceptual phase redundancy in speech

引用

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

作者： Doh-Suk Kim Human & Computer Interaction Laboratory Samsung Advanced Institute of Technology Yongin si Kyunggi South Korea

For efficient quantization of speech representation, it is common to incorporate perceptual characteristics of human hearing. However, the focus has been confined only to the magnitude information of speech, and little attention has been paid to phase information. This paper presents a novel approach, termed perceptually irrelevant phase elimination (PIPE), to find out irrelevant phase information of acoustic signals in terms of perception. The proposed method, which is based on the observation that the relative phase relationship within a critical band is perceptually important, is derived not only for stationary Fourier signal but also for harmonic signal. The proposed method is then incorporated into the analysis-by-synthesis system based on harmonic representation of speech, and subjective test results demonstrate its effectiveness.

关键词： Humans Auditory system Speech coding Quantization Speech analysis System testing Timbre Laboratories Harmonic analysis audio coding

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A new perceptual entropy-based method to achieve a signal adapted wavelet tree in a low bit rate perceptual audio coder

A new perceptual entropy-based method to achieve a signal ad...

引用

European Signal Processing Conference (EUSIPCO)

作者： N. Ruiz Reyes M. Rosa Zurera F. López Ferreras D. Martínez Muñoz Departamento de Electrónica Universidad de Jaén Escuela Universitaria Politénica. 23700 Linares - Jaén (Spain) Dpto. de Teoría de la Señal y Comunicaciones Universidad de Alcalá Escuela Politécnica 28871 Alcalá de Henares - Madrid (Spain)

This paper outlines a wavelet based audio coding scheme that uses a perceptual entropy criterion to achieve a signal adapted wavelet tree. We are interested in decompositions adapted to the nature of audio signals, but attending to the characteristics of the human hearing system. Our audio coder is compared with two other wavelet based coding schemes: the first one implements a fixed wavelet decomposition which closely mimics the critical band division at the inner ear, while the second one uses an adaptive wavelet tree controlled by a classical entropy criterion, such as Shannon entropy. The results confirm that perceptual entropy-based decomposition is the way you must take if you are looking for maximum compression rates and transparent coding. To achieve a further bit rate reduction ensuring transparent coding, our audio coder performs a filter order optimization, which is possible by using a novel algorithm to translate the psycho-acoustic information from the frequency to the wavelet domain. Experimental results indicate that the proposed coder ensures transparent coding of monophonie CD quality audio signals at bit rates lower than 64 kb/s (1.45 bit/sample) for most audio signal. Subjective listening tests tell us that our coding method behaves as well as ISO/MPEG 1 — layer III and much better than MPEG 2-layer II at that bit rate.

关键词： Bit rate Wavelet packets Channel coding Cost function audio coding Transform coding

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MPEG-4 AAC audio decoding on a 24-bit fixed-point dual-DSP architecture

MPEG-4 AAC audio decoding on a 24-bit fixed-point dual-DSP a...

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： V. Mesarovic N.D. Hemkumr M. Dokic Crystal Audio Division Cirrus Logic Corporation Austin TX USA

MPEG-4 audio coding is a recent audio standard that integrates a wide range of different types of audio coding, and, thus, applies to almost every audio application today. A very important subpart of MPEG-4 is the Advanced audio coding (AAC) algorithm, which has already distinguished itself in industry due to its high audio quality. This paper describes implementation of an AAC decoder on a novel dual-DSP architecture utilizing cost-effective 24-bit fixed-point arithmetic.

关键词： MPEG 4 Standard Decoding audio coding Digital signal processing IEC standards ISO standards Digital audio broadcasting Speech synthesis HDTV Speech coding

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