• Proceedings of the IEEK Conference
• /
• 2004.06b
• /
• pp.399-402
• /
• 2004

This paper presents a system-on-chip prototype implementing a full integration for a portable digital audio system. The chip is composed of a audio processor block to implements audio decoding and voice compression or decompression software, a system control block including 8-bit MCU core and Memory Management Unit (MMU) a low power 16-bit ${\Sigma}{\Delta}$ CODEC, two DC-to-BC converter, and a flash memory controller. In order to support other audio algorithms except Mask ROM type's fixed codes, a novel 16-bit fixed-point DSP core with the program-download architecture is proposed. Funker, an efficient power management technique such as task-based clock management is implemented to reduce power consumption for portable application. The proposed chip has been fabricated with a 4 metal 0.25um CMOS technology and the chip area is about 7.1 mm ${\times}$ 7.1mm with 100mW power dissipation at 2.5V power supply.

• Proceedings of the IEEK Conference
• /
• 2003.07d
• /
• pp.1419-1422
• /
• 2003

Multimedia contents what we call Digital Items include various types of resources such as music, speech, text, video, graphics, and so on. The current Adaptation QoS described in the ISO/IEC 21000-7 CD-Part 7: Digital Item Adaptation, however, lacks adaptation methods for audio signals. The goal of this paper is to provide adaptation methods that are necessary to deal with audio signals. Two operations are introduced in order to adapt audio items. One method is to make use of the functionality of Fine Grain Scalability, and the other is intended to drop the channel of audio output channel. This paper provides a DIA description tool that associates the operators with the corresponding values of the constraint and the utility. Furthermore, the operations are evaluated and compared to alternative solutions.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.12 no.2
• /
• pp.662-675
• /
• 2018

Compression history detection plays an important role in digital multimedia forensics. Most existing works, however, mainly focus on digital image and video. Additionally, the existed audio compression detection algorithms aim to detect the trace of double compression. In real forgery scenario, multiple compression is more likely to happen. In this paper, we proposed a detection algorithm to reveal the compression history for MP3 audio. The statistics of the scale factor and Huffman table index which are the parameters of MP3 codec have been extracted as the detecting features. The experimental results have shown that the proposed method can effectively identify whether the testing audio has been previously treated with single/double/triple compression.

• Proceedings of the KIEE Conference
• /
• 1995.07b
• /
• pp.977-979
• /
• 1995

MPEG-II is the international standard of compression for digital image and digital audio that is the most important in the multimedia environment. Now many researchers are developing relevant systems. MPEG-II consists of video, audio, system and the other part. In this paper, we have designed and demonstrated two channel audio encoder system that processes the audio compression part, and excutes layer II for complexity and psychoacoustic model II, with ADSP-21020 of Analog Device.

• Journal of Broadcast Engineering
• /
• v.6 no.1
• /
• pp.13-20
• /
• 2001

In this paper, we propose a new digital audio watermarking scheme In the cepstrum domain. We insert a digital watermark signal Into the cepstral components of the audio signal using a technique analogous to spread spectrum Communications, hiding a narrow band signal in a wade band channel. In our proposed method, we use pseudo-random sequences to watermark the audio signal. The watermark Is then weighted in the cepstrum domain according to the distribution of cepstral coefficients and the frequency masking characteristics of the human auditory system. The proposed watermark embedding scheme minimizes audibility of the watermark signal. and the embedded watermark is robust to mu1tip1e watermarks, MPEG audio ceding and additive noose.

• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 2000.11b
• /
• pp.43-48
• /
• 2000

This paper introduces MPEG-2 AAC codec system fur digital audio broadcasting. This system consists of encoder and decoder, and this system provides MPEG-2 system multiplexing and demultiplexing functions. Four DSPs are adopted fur encoder and three DSPs fur decoder. Each DSP processes system control, I/O control, and audio signal processing, multiplexing and demultiplexing. This paper also discusses about some near future estimations related to DAB system and services. And at the end of this paper describes about future development plans.

• Proceedings of the IEEK Conference
• /
• 2003.11b
• /
• pp.243-246
• /
• 2003

In this paper, we propose a new digital audio watermarking technique with the wavelet transform. The watermark is embedded by eliminating unnecessary information of audio signal based on human auditory system (HAS). This algorithm is an audio watermarking method, which does not require any original audio information in watermark extraction process. In this paper, the masking effect is used for audio watermarking, that is, post-tempera] masking effect. We construct the window with the synchronization signal and we extract the best frame in the window by using the zero-crossing rate (ZCR) and the energy of the audio signal. The watermark may be extracted by using the correlation of the watermark signal and the portion of the frame. Experimental results show good robustness against MPEG1-layer3 compression and other common signal processing manipulations. All the attacks are made after the D/A/D conversion.

• Journal of Broadcast Engineering
• /
• v.11 no.2 s.31
• /
• pp.181-190
• /
• 2006

Technology for compressing low-bitrate multichannel audio coding should be developed owing to the increasing need of consumer for multichannel audio contents and services. To meet this requirement, MPEG has standardized MPEG Surround. In this paper, we introduce status on MPEG Surround standardization and analyze techniques adopted in the current MPEG Surround.

• JSTS:Journal of Semiconductor Technology and Science
• /
• v.14 no.1
• /
• pp.109-116
• /
• 2014

This paper presents a high-efficiency digital class-D audio amplifier using a composite interpolation filter for portable audio devices. The proposed audio amplifier is composed of an interpolation filter, a delta-sigma modulator, and a class-D output stage. To reduce power consumption, the designed interpolation filter has an optimized composite structure that uses a direct-form symmetric and Lagrange FIR filters. Compared to the filters with homogeneous structures, the hardware cost and complexity are reduced by about half by the optimization. The coefficients of the digital delta-sigma modulator are also optimized for low power consumption. The class-D output stage has gate driver circuits to reduce shoot-through current. The implemented class-D audio amplifier exhibited a high efficiency of 87.8 % with an output power of 57 mW at a load impedance of $16{\Omega}$ and a power supply voltage of 1.8 V. An outstanding signal-to-noise ratio of 90 dB and a total harmonic distortion plus noise of 0.03 % are achieved for a single-tone input signal with a frequency of 1 kHz.

• Proceedings of the IEEK Conference
• /
• 2000.06d
• /
• pp.110-113
• /
• 2000

In this paper, we consider the problem of digital audio watermarking to robust about compression without original audio data. We specifically address the audio watermarking using BPSK with variable carrier frequency. This technique make audio data embeded watermarking robust with compression attack, for example MPEG, AC-3, etc.