• The Journal of the Acoustical Society of Korea
• /
• v.30 no.6
• /
• pp.337-343
• /
• 2011

With commencing trial 3D digital broadcasting, the studies on next generation digital broadcasting technology for coming UHDTV era is being actively progressing. In this paper, I propose surround audio formats for next-generation digital TV broadcasting, along with comparative study of major surround audio formats in use or under development. I did comparative study on current major competing surround formats such as Dolby True HD and DTS HD MA, along with NHK proposed 22.2 channel surround format for UHDTV system. Upon this comparative study and our housing situation consideration, I propose lossy compression 3D surround 7.1 channel surround format along with loosless 2.0 and 4.0 hi-fi format as next generation digital TV broadcasting standard. In lieu with this, I also propose transmitting binaural 2 channel audio data as sub-audio. It will give holographic sound experience when properly processed with individual HRTF (Head Related Transfer Function) with headphone. The table for data rate of each proposed audio format is also presented.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.29 no.6C
• /
• pp.865-873
• /
• 2004

MPEG-2 Advanced Audio Coding(AAC) is widely used in the multi-channel audio compression standards. And it combines hi인-resolution filter bank prediction techniques, and Huffman coding algorithm to achieve the broadcast-quality audio level at very low data rates. The forward and inverse modified discrete transforms which are operated in the encoder and the decoder of the filter bank need many computations. In this paper, we propose suitable recursive structure at IMDCT processing for MPEG-2 AAC real-time decoder. We confirm the memory, the computation speed and complexity of the proposed structure.

• ETRI Journal
• /
• v.25 no.1
• /
• pp.52-54
• /
• 2003

We propose an improved algorithm for detecting audio watermarks based on a spread spectrum in the spectral domain. Since the energy of a watermark is much smaller than that of the cover audio data, pre-processing to reduce the effect of the cover data is needed to reliably extract watermarks. We introduce a spectral envelope filter as a pre-process that enhances detecting performance by filtering out the intrinsic spectral character of cover data. The proposed watermarking structure can be easily included in the compression system and can extract watermarks from partially decompressed spectral data. Our experimental results demonstrate that with a bit error rate of around 10 dB against general attacks, the proposed detecting scheme works better than detectors without the spectral filter.

• Proceedings of the IEEK Conference
• /
• summer
• /
• pp.114-117
• /
• 2004

This paper proposes a novel algorithm for hybrid lossless audio coding, which employs integer discrete cosine transform. The proposed algorithm divides the input signal into frames of a proper length, decorrelates the framed data using the integer DCT and finally entropy-codes the frame data. In particular, the adaptive Golomb-Rice coding method used for the entropy coding selects an optimal option which gives the best compression efficiency. Since the proposed algorithm uses integer operations, it significantly improves the computation speed in comparison with an algorithm using real or floating-point operations. When the coding algorithm is implemented in hardware, the system complexity as well as the power consumption is remarkably reduced. Finally, because each frame is independently coded and is byte-aligned with respect to the frame header, it is convenient to move, search, and edit the coded data.

• Proceedings of the IEEK Conference
• /
• 2002.07c
• /
• pp.2063-2066
• /
• 2002

In this paper, we propose a structure far lossless audio coding method. Prediction model is used in the wavelet transform domain. After DWT, wavelet coefficients is quantized and decorrelated by prediction modeling. The DWT can be constructed to critical bands. We can get a lower data rate representation of audio signal which has a good quality like the result of perceptual coding. Then the prediction errors are efficiently coded by the Golomb-coding method. The prediction coefficients are fixed for reducing the computational burden when we find prediction coefficients.

• Journal of the Korean Institute of Telematics and Electronics C
• /
• v.35C no.3
• /
• pp.59-68
• /
• 1998

This paper propose a hardware architecure of multimediasysgem for integrated processing of the multimedia data such as audio and video, and describes on the design and implementation of multimedia integrated processing Unit. The unit comprises most commonly needed multimedia processing function for computer-based video conference: audio-visual datacapture, playback, compression, decompression as well as interleaving/disinterleaving of compressed audio-visual data. The proposed architecture minimizes the CPU overhead that might be caused by multimedia data processing and assures the fluent data flow among system components. Also, this unit is tested and analyzed under the computer-based video conference to confirm the multimedia unit of proposed architecture using communication protocol and application software through Ethernet and FDDI (Fiber Distributed Data Interface) networks.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• v.9 no.1
• /
• pp.1133-1136
• /
• 2005

The MPEG/audio standard results from three years of co-work by an international committe of high-fidelity audio compression experts in the Moving Picture Experts Group (MPEG/audio). The MPEG standard is rigid only where necessary to ensure interoperability. In this paper, a new approach of training and classification of acoustic signal is addressed. This is some what a fields of application aspects rather than technonical problems such as MPEG/codec, MIDI. In preprocessing, acoustic signal is transformmed using DWT so as to extract a feature parameters of sound such as loudness, pitch, bandwidth and harmonicity. these accoustic parameters are exploited to the input vector of neural perceptron. Experimental results showed that proposed approach can be used for tunning the dissonance chord.

• The Journal of the Acoustical Society of Korea
• /
• v.22 no.5
• /
• pp.367-374
• /
• 2003

In this paper, we propose an audio watermarking method using the attribute of tonal masker. First, the attribute of tonal masker as an audio watermarking attribute is analyzed. According to existing researches, it is possible to be imperceptible modulation for the energies of the frequencies that compose a tonal masker. And when the relation between the tone energy and the left or right frequency energy after various signal processing is compared with the one before the processing, very few changes are showed. We propose an audio watermarking method using these attributes of tonal masker. A watermark bit is embedded by the modulation of the difference between the two neighboring frequency energies of a tone. In the detection, the modulated the tonal masker is searched using the key wed in the embedding without original audio and the embedded watermark bit is detected. After each attack of noise insertion, band-pass filtering, re-sampling, compression, echo transform and equalization, the detection error ratios of the proposed method were average 0.11%, 1.26% for Classics and Pops. And the SDG(Subjective Diff-Grades) scale evaluation of the sound quality of the watermarked audio result in the average SDG -0.31.

• 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.

• 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.