• Journal of the Institute of Electronics Engineers of Korea SD
• /
• v.39 no.5
• /
• pp.65-73
• /
• 2002

In this paper, we propose and verify an implementation method for a single-chip 2048 complex point FFT/IFFT in terms of sequential data processing. For the sequential processing of 2048 complex data, buffers to store the input data are necessary. Therefore, DRAM-like pipelined commutator architecture is used as a buffer. The proposed structure brings about the 60% chip size reduction compared with conventional approach by using this design method. The 16-point FFT is a basic building block of the entire FFT chip, and the 2048-point FFT consists of the cascaded blocks with five stages of radix-4 and one stage of radix-2. Since each stage requires rounding of the resulting bits while maintaining the proper S/N ratio, the convergent block floating point (CBFP) algorithm is used for the effective internal bit rounding and their method contributed to a single chip design of digital audio broadcasting system.

• Journal of Korea Society of Industrial Information Systems
• /
• v.19 no.2
• /
• pp.73-83
• /
• 2014

Thanks to the development of diverse audio editing Technology, audio file can be easily revised. As a result, diverse social problems like forgery may be caused. Digital forensic technology is actively studied to solve these problems. In this paper, a hand-held device identification method, an area of digital forensic technology is proposed. It uses the noise features of devices caused by the design and the integrated circuit of each device but cannot be identified by the audience. Wiener filter is used to get the noise sounds of devices and their acoustic features are extracted via MIRtoolbox and then they are trained by multi-layer neural network. To evaluate the proposed method, we use 5-fold cross-validation for the recorded data collected from 6 mobile devices. The experiments show the performance 99.9%. We also perform some experiments to observe the noise features of mobile devices are still useful after the data are uploaded to UCC. The experiments show the performance of 99.8% for UCC data.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
• /
• v.13 no.2
• /
• pp.176-187
• /
• 2002

The digital Audio Broadcasting (DAB) system adopts Coded OFDM(COFDM) for channel coding. The COFDM is a combined technique of multicarrier transmission(OFDM) and punctured convolutional coding with viterbi error correction. Because the channel coding is an important topic for OFDM systems, this paper proposes a new turbo coded OFDM system that replaces the existing RCPC codec by a turbo codec without modifying the puncturing procedure and puncturing vectors defined in the standard DAB system for compatibility. The performance of a new system is compared to that of the conventional system under the frequency selective Rician fading channel and the frequency selective Rayleigh fading channel in conjunction with DAB transmission mode I suitable for the terrestrial single frequency network(SFN) broadcasting. The standard system's performance was improved with the aid of turbo codec.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.29 no.6C
• /
• pp.874-882
• /
• 2004

MPEG Audio compression algorithm is the international standard for the digital compression of high quality audio using mechanism of the perceptual coding based on psychoacoustic masking. It is necessary to discuss the constraints on designing of common filter banks for MPEG-2 BC and MPEG-2 AAC decoder system, which is not Down yet, mapping audio signals from the time domain into the frequency domain. In this paper, we present an architecture of common synthesis filter whcih can be used for MPEG-2 BC and MPEG-2 AAC decoder using recursive structure. The proposed algorithm is based on recursive architecture that effectively performs common compulsion.

• Proceedings of the IEEK Conference
• /
• 2004.08c
• /
• pp.477-481
• /
• 2004

High Quality CD, OAT audio requires that large amount of data. Currently, multi channel preference has been rapidly propagated among latest users. The MPEG(Moving Picture Expert Group) is provides data compression technology of sound and image system. The MPEG standard provides multi channel and 5.1 sounds, using the same audio algorithm as MPEG-l. And MPEG-2 audio is forward and backward compatible. The MDCT (Modified Discrete Cosine Transform) is a linear orthogonal lapped transform based on the idea of TDAC(Time Domain Aliasing Cancellation). In this paper, we proposed the micro-program type vector processor architecture a benefit in MDCT/IMDCT of MPEG-II AAC. And it's reduced operating coefficient by overlapped area to bind. To compare original algorithm with optimized algorithm that cosine coefficient reduced $0.5\%$multiply operating $0.098\%$ and add operating 80.58\%$. Algorithm test is used C-language then we designed hardware architecture of micro-programmed method that applied to optimized algorithm. This processor is 20MHz operation 5V.

• The KIPS Transactions:PartB
• /
• v.11B no.5
• /
• pp.587-596
• /
• 2004

During recent years, a new framework, which aims to bring a unified and global approach in indexing, browsing and querying various digital multimedia data such as audio, video and image has been developed. This new system partitions each media stream into smaller units based on actual physical events. These physical events within oath media stream can then be effectively indexed for retrieval. In this paper, we present a new approach that exploits audio, image and video features to segment and analyze the audio-visual data. Integration of audio and visual analysis can overcome the weakness of previous approach that was based on the image or video analysis only. We Implement a web-based multi media data retrieval system called XCRAB and report on its experiment result.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.36 no.10B
• /
• pp.1252-1257
• /
• 2011

In this paper, efficient functions for audio spectrum mapping with visible spectrum are proposed. Through mapping overall hearing frequency band with visible frequency band, emotional lighting might be possible. We propose a basic linear mapping function and non-linear mapping functions emphasizing specific audio frequency bands. For the algorithm implementation, spectrum analysis method and filter method are introduced. Especially, in this paper, a prototype LED lighting equipment using the digital filter method is implemented. The proposed lighting method can be applied to many LED lighting area using music.

• Journal of the Institute of Convergence Signal Processing
• /
• v.15 no.1
• /
• pp.1-8
• /
• 2014

Digital watermarking techniques are attracting attention as a proper solution to protect copyright for multimedia data. This paper proposes a new audio watermarking method based on Discrete Cosine Transformation (DCT) and Discrete Wavelet Transformation (DWT) for copyright protection. In our proposed watermarking method, the original audio is transformed into DCT domain and divided into two parts. Synchronization code is applied on the signal in first part and 2 levels DWT domain is applied on the signal in second part. The absolute value of DWT coefficient is divided into arbitrary number of segments and calculates the energy of each segment and middle peak. Watermarks are then embedded into each middle peak. Watermarks are extracted by performing the inverse operation of watermark embedding process. Experimental results show that the hidden watermark data is robust to re-sampling, low-pass filtering, re-quantization, MP3 compression, cropping, echo addition, delay, and pitch shifting, amplitude change. Performance analysis of the proposed scheme shows low error probability rates.

• Journal of the Korean Institute of Telematics and Electronics
• /
• v.26 no.7
• /
• pp.90-97
• /
• 1989

In this paper, we have taken the formation of two-dimension codeword named doubly-encoded code using the Reed-Solomon code, C1(32, 28) with minimum distance 5 and C2(32, 26) with minimum distance 7 and we have had computer simulation of these error correcting processes using modeled R-DAT (Rotationary Digital Audio Tape). As the result, the error rate per symbol has been decreased about 0.05 and on these processes, the newly developed digital signal processing technology such as erro correction using Berlekamp-Massey algorithm in frequency domain have been proven.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.16 no.2
• /
• pp.253-258
• /
• 2012

A computationally efficient interpolation filter with a low-distortion performance is a key component to utilize the naturally-sampled pulse width modulation (NPWM) in a digital domain. To realize the efficient interpolation filter, we propose a novel design based on the recently-proposed modified Farrow filter. The proposed filter shows a better pass-band distortion performance maintaining similar degree of complexity compared with the conventional Lagrange interpolation filter. We achieve the maximum distortion deviation of 10-3 dB to 20-kHz audible frequency range and distortion reduction of 1/6 times compared with the Lagrange interpolation filter.