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Design and implementation of filterbank for MPEG-2/4 AAC system

INTEGRATION-THE VLSI JOURNAL 2022年 82卷 155-162页

作者： Tsai, Tsung-Han Liu, Hsing-Chuang Natl Cent Univ Dept Elect Engn Jhongli 320 Taoyuan County Taiwan

In this paper, the design and implementation of low power and high-efficiency filterbank for MPEG-2/4 AAC system is presented. Since filterbank represents the most computation-intensive kernel of AAC codec, we design it with algorithm and architecture aspects. We supply the dedicated algorithm for filterbank. The derived algorithm and the hardware shared engine (HSE) we proposed for AAC can reduce the computation power and implement as a codec used in encoder and decoder. We also optimize the performance, hardware resources, and power consumption thoroughly. It is designed as an intellectual property (IP) to construct the overall decoder in an embedded system. The hardware cost is with 16.1 k logic gates, 2 k-word local memory, and 1 K-word coefficient ROM. The proposed design has a real-time operation at only 1.25 MHz with a sampling rate of 48 kHz. It can achieve 0.70 mW power consumption in TSMC 0.18 mu m CMOS technology. Furthermore, we use a programmable chip (SOPC) platform which includes the software and hardware engine. Throughout the computation analysis, the bitstream parser and lower complexity part are performed by a software solution, and the higher complexity part is computed by a hardware solution. Several design techniques are needed including the wrapper design, embedded CPU, and IP to construct the system. Based on this co-design approach, a whole AAC audio decoder is also established.

关键词： digital audio coding MPEG-2 4 AAC codec Filterbank MDCT IMDCT IP SOPC HW SW co-Design

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Keynote 1: "Recent developments in audio technology"

Keynote 1: "Recent developments in audio technology"

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IEEE International Conference on Consumer Electronics - Berlin (ICCE-Berlin)

作者： Karlheinz Brandenburg Fraunhofer Institute for Digital Media Technology TU Ilmenau DE

Dr. Karlheinz Brandenburg has been a driving force behind some of today's most innovative digital audio technology, notably the MP3 and MPEG audio standards. He is acclaimed for seminal work in digital audio coding and perceptual measurement techniques, Wave Field Synthesis (WFS) and psycho-acoustics. Karlheinz Brandenburg is a full professor at the Institute for Media Technology at Technische Universität Ilmenau. At the same time he is the director of the Fraunhofer Institute for digital Media Technology IDMT in Ilmenau.

关键词： Brandenburg Germany Research Institutes digital audio coding audio coding digital audio WFS1 gene DIRECTORS digital media digital audio players helium Moving Pictures Experts Group DRIVING FORCE

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A VLSI Implementation of Filterbank for MPEG2/4 AAC Encoder

A VLSI Implementation of Filterbank for MPEG2/4 AAC Encoder

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International Conference on Applied Materials and Electronics Engineering (AMEE 2012)

作者： Tsai, Tsung-Han Sha, Yu-Jie Natl Cent Univ Dept Elect Engn Jhongli 32001 Taiwan

ISBN: (纸本)9783037852880

In this paper, we proposed a VLSI architecture of the modified discrete cosine transform (MDCT) for MPEG 2/4 AAC encoders. The MDCT transforms the time domain input signals to the frequency domain spectrums. It is considered one of the most computational intensive part in implementing the AAC encoder. The AAC encoder allows four types of audio blocks while encoding the audio files. With our algorithm, the proposed design can share the coefficients needed in the MDCT. Moreover, we used a 2-dimensional folding technique to reduce the hardware cost while maintaining the audio quality. The proposed design is realized in TSMC 0.18-um 1P6M technology and is operated at 50 MHz. With these techniques on special hardware design, the result shows some advantages on low complexity.

关键词： digital audio coding MPEG-2/4 AAC encoder filterbank MDCT

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Introduction to AVS audio

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Journal of Computer Science & Technology 2006年第3期21卷 360-365页

作者：艾浩军陈水仙胡瑞敏 National Engineering Research Center for Multimedia Software Wuhan University Wuhan 430079 P.R. China Hubei Multimedia Network Communication Engineering Laboratory Wuhan University Wuhan 430079 P.R. China Computer School Wuhan University Wuhan 430072 P.R. China Computer School Wuhan University Wuhan 430072 P.R. China Computer School Wuhan University Wuhan 430072 P.R. China Computer School Wuhan University Wuhan 430072 P.R. China

This paper describes a general audio coding algorithm which has been recently standardized by AVS, China. The algorithm is based on a perceptual coding technique. The codec delivers near CD-quality audio at 128kb/s. This paper describes the coder structure in detail and discusses the reasons for specific design methods. A summary of the subjective test results are presented for the prototype codec. Comparison Mean Opinion Score （CMOS） test indicates that the quality of the AVS audio coder is comparable with MPEG Layer-3 audio coder. A reM-time decoder was used for the characterization test, which is based on a 16-bit fixed-point DSP. The performance of the DSP solution was demonstrated, including computational complexity and storage characteristics.

关键词： digital audio coding stereo hi-fidelity

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An integrated error correction and detection system for digital audio broadcasting

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IEEE TRANSACTIONS ON BROADCASTING 2000年第1期46卷 68-78页

作者： Chen, B Sundberg, CEW Bell Labs Lucent Technol Multimedia Commun Res Lab Murray Hill NJ 07974 USA

Hybrid in-band on-channel digital audio broadcasting systems deliver digital audio signals in such a way that is backward compatible with existing analog FM transmission. We present a channel error correction and detection system that is well-suited for use with audio source coders, such as the so-called perceptual audio coder (PAC), that have error concealment/mitigation capabilities. Such error mitigation is quite beneficial for high quality audio signals. The proposed system involves an outer cyclic redundancy check (CRC) code that is concatenated with an inner convolutional code. The outer CRC code is used for error detection, providing flags to trigger the error mitigation routines of the audio decoder, The inner convolutional code consists of so-called complementary punctured-pair convolutional codes, which are specifically tailored to combat the unique adjacent channel interference characteristics of the FM band. We introduce a novel decoding method based on the so-called list Viterbi algorithm (LVA). This LVA-based decoding method, which may be viewed as a type of joint or integrated error correction and detection, exploits the concatenated structure of the channel code to provide enhanced decoding performance relative to decoding methods based on the conventional Viterbi algorithm (VA). We also present results of informal listening tests and other simulations on the Gaussian channel. These results include the preferred length of the outer CRC code for 96-kb/s audio coding and demonstrate that LVA-based decoding can significantly reduce the error nag rate relative to conventional VA-based decoding, resulting in dramatically improved decoded audio quality. Finally, we propose a number of methods for screening undetected errors in the audio domain.

关键词： DAB IBOC hybrid IBOC digital audio coding PAC channel coding list Viterbi algorithm error concealment

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Multistream hybrid in band on channel FM systems for digital audio broadcasting

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IEEE TRANSACTIONS ON BROADCASTING 1999年第4期45卷 410-417页

作者： Sundberg, CEW Sinha, D Mansour, D Zarrabizadeh, M Laneman, JN Lucent Technol Bell Labs Multimedia Commun Res Lab Murray Hill NJ 07974 USA Lucent Technol Lucent Digital Radio Warren NJ 07059 USA Lucent Technol Bell Labs Holmdel NJ 07933 USA

New approaches to hybrid in band on channel (HIBOC) FM systems for digital audio broadcasting based on multistream transmission methodology and multidescriptive audio coding techniques are introduced in this paper. These ideas combined with a lower per sideband audio coding rate and more powerful channel codes result in robust transmission and graceful degradation in variable interference channels. By also using orthogonal frequency division multiplexing techniques with a nonuniform power profile combined with unequal error protection and sideband time diversity, we arrive at new HIBOC FM schemes with extended coverage and better peak audio quality than previously proposed. The paper provides approximate performance analysis for potential systems including audio coding quality.

关键词： DAB IBOC hybrid IBOC multistream transmission PAC digital audio coding

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