• Journal of Information Processing Systems
• /
• v.12 no.3
• /
• pp.525-537
• /
• 2016

Data hiding is a wide field that is helpful to secure network communications. It is common that many data hiding researchers consider improving and increasing many aspects such as capacity, stego file quality, or robustness. In this paper, we use an audio file as a cover and propose a reversible steganographic method that is modifying the sample values using modulus function in order to make the reminder of that particular value to be same as the secret bit that is needed to be embedded. In addition, we use a location map that locates these modified sample values. This is because in reversible data hiding it needs to exactly recover both the secret message and the original audio file from that stego file. The experimental results show that, this method (measured by correlation algorithm) is able to retrieve exactly the same secret message and audio file. Moreover, it has made a significant improvement in terms of the following: the capacity since each sample value is carrying a secret bit. The quality measured by peak signal-to-noise ratio (PSNR), signal-to-noise ratio (SNR), Pearson correlation coefficient (PCC), and Similarity Index Modulation (SIM). All of them have proven that the quality of the stego audio is relatively high.

• The Journal of the Acoustical Society of Korea
• /
• v.15 no.2
• /
• pp.55-60
• /
• 1996

This paper describes the subjective assessment of the multi-channel audio quality according to the bitrate changing and evaluates the predistortion effect to avoid the unmasked noise after matrixing/dematrxing process in transmission and regeneration of the multi-channel audio. The simulation is processed by the perceptual coding that is MPEG-2 Audio layer II algorithm. We evaluate the quality improvement about predistortion using or not by 384, 320, 256, 128kbps. As the result of the double blind subjective assessment, 5 Grade-Impairment Scale is scored under minus one to 320kbps and so audio quality is evaluated to be perceptible, but not annoying in 3/2 channel. The effect of the predistortion is improved one level in 128kbps and especially speech test material I better improved than music test materials.

• Proceedings of the IEEK Conference
• /
• 2002.07c
• /
• pp.1567-1570
• /
• 2002

This paper deals with FPGA(Field Programmable Gate Array) implementation of the AAC(Advanced Audio Coding) decoder. On modern computer culture, according to the high quality data is required in multimedia systems area such as CD, DAT(Digital Audio Tape) and modem. So, the technology of data compression far data transmission is necessity now. MPEG(Moving Picture Experts Group) would be a standard of those technology. MPEG-2 AAC is the availableness and ITU-R advanced coding scheme far high quality audio coding. This MPEG-2 AAC audio standard allows ITU-R 'indistinguishable' quality according to at data rates of 320 Kbit/sec for five full-bandwidth channel audio signals. The compression ratio is around a factor of 1.4 better compared to MPEG Layer-III, it gets the same quality at 70％ of the titrate. In this paper, for a real time processing MPEG2 AAC decoding, it is implemented on FPGA chip. The architecture designed is composed of general DSP(Digital Signal Processor). And the Processor designed is coded using VHDL language. The verification is operated with the simulator of C language programmed and ECAD tool.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.11 no.8
• /
• pp.1450-1455
• /
• 2007

Recently DSP chips are widely used in high quality car audio to achieve high quality sound and to process various sound sources, e.g. navigation, cell phones. In this paper, we developed a set of design tools useful for developing high quality car audio systems using Philips' SAF7730 car audio chips. The tool is consist of the GUI(Graphic User Interface) program running on the Windows operating system and the interface board which performs data conversion between RS232C and I2C protocols. The developed system has been successfully applied to commecial car audio design.

• Journal of Broadcast Engineering
• /
• v.14 no.3
• /
• pp.332-341
• /
• 2009

Terrestrial digital multimedia broadcasting(T-DMB) is one of mobile broadcasting services, and the commercial service was started in December 2005 in Korea. The performance targets of T-DMB are providing VCD(video CD) quality video and FM radio quality audio. In recent years, the researches for providing high quality video or audio service on T-DMB environments have been being carried out. To provide high-quality video or audio service, some additional data should be transmitted to the receiver as well as T-DMB video and audio data. Since the data rate for one T-DMB program is low, it is important to transmit the additional data at a low bit rate. In this paper, we propose a packetizing method for efficient transmission of the additional data to provide multichannel audio service on T-DMB environment.

• Journal of the Institute of Electronics and Information Engineers
• /
• v.50 no.12
• /
• pp.232-237
• /
• 2013

The role of the window length of time-frequency transformation is important for the audio data hiding methods utilizing phase information. In this paper, the experiments for our audio data hiding method were conducted in order to evaluate the audio quality and robustness against reverberant environment. The experimental results showed the tendency that the worse audio quality but better robustness were obtained when the lengthy window was applied. The important reason for quality degradation was pre-echo which flatters the percussive sound. The results also indicated that the wireless communication theory related to the length of time-frequency transform can be applied in the field of audio data hiding and acoustic data transmission.

• Proceedings of the IEEK Conference
• /
• 2006.06a
• /
• pp.351-352
• /
• 2006

The objective criterion is necessary to decide the signal for vehicle audio. In this time, the signals have decided by golden ear or some simple parameters. In this paper, we propose the system for sound quality assessment in the vehicle audio signals. The vehicle audio signals were recorded in many vehicles by same condition. From these signals, objective and subjective parameters were extracted and using these signals listening test was done. The function was estimated between three results by simple neural network system. The score of arbitrary signal can be estimated using the function made by three results. In this way, Sound signal quality is assessed objectively.

• Journal of Korea Multimedia Society
• /
• v.7 no.1
• /
• pp.54-63
• /
• 2004

This paper describes a hybrid QoS guarantee scheme for high quality audio streaming services on the Internet. The continuous playback of the audio data requires the isochronous transmission of the audio data packet through the Internet. In order to retain the QoS at the ultimate destination (client) as the same as servers provide, the transmission protocols should consider the error conditions such as packet loss, and out of order delivery. Generally, the protocols supporting the transmission of continuous media data do not try to recover the errors. The protocols are working somehow for the toll quality multimedia streaming services, but rot for the high quality streaming services, such as the DVD sound/music payback. The hybrid QoS guarantee scheme includes the three mechanisms to overcome the problem. The selective retransmission for the lost packet, the adaptive buffering at client-side, and the adaptive transmission rate at server-side are totally adopted to recover the packet loss with the minimal overhead, to prevent from the buffer starvation during the retransmission, and to maintain the isochronous transmission even after the retransmission. The experiments have shown good results for the high Quality audio streaming services on the Internet.

• The Journal of the Acoustical Society of Korea
• /
• v.23 no.3E
• /
• pp.74-78
• /
• 2004

As analog audio systems are changing to digital systems, the DAB (Digital Audio Broadcasting) is expected to provide CD quality audio, various data services with interactiveness and excellent mobile reception ability. The DMB (Digital Multimedia Broadcasting), as more advanced successor of the DAB, adds video capability on the audio and data services. The DAB system assures high quality audio services even when the reception is through portable and mobile receivers. In this paper, USB-DAB receiver and PCI-DMB receiver are designed and implemented. The DAB receiver and the DMB receiver incorporate with PC to make use of computational power and application software of Pc. This enables the developed system to be more flexible and to meet various applications easier.

• Proceedings of the KSPS conference
• /
• 2006.11a
• /
• pp.131-134
• /
• 2006

Quality of narrowband speech $(0{\sim}4kHz)$ can be enhanced by the bandwidth expansion technique, by which the high- band components are estimated. This paper proposes the bandwidth expansion method using the spline codebook based spectral folding. For the performance evaluation, the PESQ(Perceptual Evaluation of Speech Quality) scores are measured as the objective measurement In addition, the MOS (Mean Opinion Score) and the preference tests are performed as the subjective measurement. The results show our proposed method outperforms the existing spline based one.