• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.3
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• pp.87-95
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• 2010

Portable multi-media products which can service the highest audio-quality by using lossless audio codec has been released and the international lossless codecs, MPEG-4 audio lossless coding(ALS) and MPEG-4 scalable lossless coding(SLS), were standardized by MPEG in 2006. The simple profile of MPEG-4 ALS, it supports up to stereo, was defined by MPEG in 2009. The lossless audio codec should have low-complexity in stereo to be widely used in portable multi-media products. But the previous researches of MPEG-4 ALS have focused on an improvement of compression ratio, a complexity reduction in multi-channels coding, and a selection of linear prediction coefficients(LPCs) order. In this paper, the complexity and compression ratio of MPEG-4 ALS encoder is analyzed in simple profile of MPEG-4 ALS, the method to reduce a complexity of MPEG-4 ALS encoder is proposed. Based on an analysis of complexity of MPEG-4 ALS encoder, the complexity of short-term prediction filter of MPEG-4 ALS encoder is reduced by using the low-complexity filter that is proposed in previous research to reduce the complexity of MPEG-4 ALS decoder. Also, we propose a joint coding decision method, it reduces the complexity and keeps the compression ratio of MPEG-4 ALS encoder. In proposed method, the operation of joint coding is decided based on the relation between cross-correlation of residual and compression ratio of joint coding. The performance of MPEG-4 ALS encoder that has the method and low-complexity filter is evaluated by using the MPEG-4 ALS conformance test file and normal music files. The complexity of MPEG-4 ALS encoder is reduced by about 24% by comparing with MPEG-4 ALS reference encoder, while the compression ratio by the proposed method is comparable to MPEG-4 ALS reference encoder.

• Proceedings of the KIEE Conference
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• 1998.07g
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• pp.2262-2264
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• 1998

This paper propose a new method for the discrete signal : Discrete Wavelet Transform(DWT). This paper is a brief introduction to the DWT and applies the DWT coding for the audio data as an example. We can have a number of hint about the compression algorithm of multimedia resources and the high performance of transmission and storage.

• Journal of Korea Multimedia Society
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• v.8 no.7
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• pp.882-888
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• 2005

MPEG-2 Advanced Audio Coding (AAC) has already been standardized as a sophisticated next generation technology AAC provides an audio signal that has CD quality at 96-128kbps/stereo. This paper describes a high-quality and efficient software implementation of an MPEG-2 AAC LC Profile encoder. Common scalefactor and noisless coding are accelerated by $45\%$ and $27\%$, respectively, through the use of TMS320C30 instructions. The implemented encoder uses 7.5kWords of program memory, 18kWords of data ROM and 92kBytes of data RAM, respectively. The results of subjective Qualify test showed that the sound quality achieved at 96kbps/stereo was equivalent to that of MP3 at 128kbps/stereo.

• The Journal of the Acoustical Society of Korea
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• v.29 no.3E
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• pp.131-139
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• 2010

This paper proposes a novel spectral hole substitution technique for low bit-rate audio coding. The spectral holes frequently occurring in relatively weak energy bands due to zero bit quantization result in severe quality degradation, especially for harmonic signals such as speech vowels. The enhanced aacPlus (EAAC) audio codec artificially adjusts the minimum signal-to-mask ratio (SMR) to reduce the number of spectral holes, but it still produces noisy sound. The proposed method selectively predicts the spectral shapes of hole bands using either intra-band correlation, i.e. harmonically related coefficients nearby or inter-band correlation, i.e. previous frames. For the bands that have low prediction gain, only the energy term is quantized and spectral shapes are replaced by pseudo random values in the decoding stage. To minimize perceptual distortion caused by spectral mismatching, the criterion of the just noticeable level difference (JNLD) and spectral similarity between original and predicted shapes are adopted for quantizing the energy term. Simulation results show that the proposed method implemented into the EAAC baseline coder significantly improves speech quality at low bit-rates while keeping equivalent quality for mixed and music contents.

• ETRI Journal
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• v.39 no.3
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• pp.310-318
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• 2017

Unified speech and audio coding (USAC) is one of the latest coding technologies. It is based on a switchable coding structure, and has demonstrated the highest levels of performance for both speech and music contents. In this paper, we propose an extended version of USAC with a single-mode of operation-which does not require a switching system-by extending the linear prediction-coding mode. The main concept of this extension is the adoption of the advantages of frequency-domain coding schemes, such as windowing and transition control. Subjective test results indicate that the proposed scheme covers speech, music, and mixed streams with adequate levels of performance. The obtained quality levels are comparable with those of USAC.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.5
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• pp.74-80
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• 2010

Speech and audio codecs have been developed based on different type of coding technologies since they have different characteristics of signal and applications. In harmony with a convergence between broadcasting and telecommunication system, international organizations for standardization such as 3GPP and ISO/IEC MPEG have tried to compress and transmit multimedia signals using unified codecs. MPEG recently initiated an activity to standardize the USAC (Unified speech and audio coding). However, USAC RM (Reference model) software has been problematic since it has a complex hierarchy, many useless source codes and poor quality of the encoder. To solve these problems, this paper introduces a new RM software designed with an open source paradigm. It was presented at the MPEG meeting in April, 2010 and the source code was released in June.

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

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.1C
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• pp.12-18
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• 2008

This paper This paper proposes a new inverse-quantization(IQ) table interpolation algorithm, specialized Digital Signal Processor(DSP) instructions and hardware architecture for digital audio codecs. Non-linear inverse quantization algorithm is representatively used in both MPEG-1 Layer-3 and MPEG-2/4 Advanced Audio Coding(AAC). The proposed instructions are optimized for the non-linear inverse quantization. The proposed algorithm can minimize operational complexity which reduces total computational load. Performance comparisons show a significant improvement of average error. The proposed instructions and hardware architecture can reduce 20% of the instruction counts and minimize computational loads of IQ algorithms effectively compared with existing IQ table interpolation algorithms. Proposed algorithm can implement commercial DSPs.

• The Journal of the Acoustical Society of Korea
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• v.16 no.4
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• pp.64-70
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• 1997

This paper is aimed to propose a new wavelet audio signal coding scheme which reduces the complexity of well-known MPEG(Moving Picture Expert Group)-Audio. The filters of MPEG0audio apply subband technique on the 16-bits PCM audio to aquire bitstream of subband sample using dynamic bit allocation. If we use the wavelet coefficients instead of subband samples and 6 bands which is less than 32 bands of MPEG-audio, the complexity can be reduced. A new audio signal compression algorithm in this paper is based on wavelet transform and the proposed algorithm is compared with MPEG-audio. At the bitrate of 256kbps, the proposed algorithm maintains the CD(Compact-disc) quality. We were able to reduce the about 40% of complexity at encoder and about 70% at decoder.

• The Journal of the Acoustical Society of Korea
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• v.29 no.1
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• pp.62-68
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• 2010

This paper proposes a new masking threshold adjustment strategy to improve the performance for speech signals in low bit-rate audio coding. After determining formant regions, the masking threshold is adjusted by using the energy ratio of each sub-band to the average energy of each formant. More quantization noises are added to the bands that have relatively large energy, but less distortion is allowed in spectral valley regions by allocating more bits, which reflects the concept of perceptual weighting widely used in speech coding. From the results of objective speech quality measure, we verified that the proposed method improves quality for the speech input signals compared to the conventional one.