• Journal of Broadcast Engineering
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• v.16 no.5
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• pp.714-721
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• 2011

MPEG audio uses the masking effect, high frequency component synthesis based on spectral band replication, and channel expansion based on parametric stereo for efficient compression of audio signals. In this paper, we present an overview of the state-of-the-art channel expansion technology in MPEG audio. We also present technical overviews and application examples to broadcasting services for HE-AAC v.2, MPEG Surround, spatial audio object coding (SAOC), and unified speech and audio coding (USAC) which are MPEG audio codecs based on the channel expansion technology.

• Journal of Broadcast Engineering
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• v.24 no.3
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• pp.387-394
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• 2019

In this study, a new method of audio high-band coding based on autoencoder with side information is proposed. The proposed method operates in the MDCT domain, and improves the performance by using additional side information consisting of the previous and current low bands, which is different from the conventional autoencoder that only inputs information to be encoded. Moreover, the side information in a time-frequency domain enables the high-band coder to utilize temporal characteristics of the signal. In the proposed method, the encoder transmits a 4-dimensional latent vector computed by the autoencoder and a gain variable using 12 bits for each frame. The decoder reconstructs the high band by applying the decoded low bands in the previous and current frames and the transmitted information to the autoencoder. Subjective evaluation confirms that the proposed method provides equivalent performance to the SBR at approximately half the bit rate of the SBR.

• Phonetics and Speech Sciences
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• v.4 no.3
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• pp.119-125
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• 2012

This paper describes a new algorithm for encoding spectral shape and envelope in the generic mode of G.718 super-wide band (SWB). In the G.718 SWB coder, generic mode coding and sinusoidal enhancement are used for the quantization of modified discrete cosine transform (MDCT)-based parameters in the high frequency band. In the generic mode, the high frequency band is divided into sub-bands and for every sub-band the most similar match with the selected similarity criteria is searched from the coded and envelope normalized wideband content. In order to improve the quantization scheme in high frequency region of speech/audio signals, the modified generic mode by the improvement of the generic mode in G.718 SWB is proposed. In the proposed generic mode, perceptual vector quantization of spectral envelopes and the resolution increase for spectral copy are used. The performance of the proposed algorithm is evaluated in terms of objective quality. Experimental results show that the proposed algorithm increases the quality of sounds significantly.

• The Journal of the Acoustical Society of Korea
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• v.21 no.6
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• pp.566-575
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• 2002

Low-titrate audio coding enables a number of Internet and mobile multimedia streaming service more efficiently. For the help of next-generation mobile telephone technologies and digital audio/video compression algorithm, we can enjoy the real-time multimedia contents on our mobile devices (cellular phone, PDA notebook, etc). But the limited available bandwidth of mobile communication network prohibits transmitting high-qualify AV contents. In addition, most bandwidth is assigned to transmit video contents. In this paper, we design a novel and simple method for reproducing high frequency components. The spectrum of high frequency components, which are lost by down-sampling, are modeled by the energy rate with low frequency band in Bark scale, and these values are multiplexed with conventional coded bitstream. At the decoder side, the high frequency components are reconstructed by duplicating with low frequency band spectrum at a rate of decoded energy rates. As a result of segmental SNR and MOS test, we convinced that our proposed method enhances the subjective sound quality only 10%∼20% additional bits. In addition, this proposed method can apply all kinds of frequency domain audio compression algorithms, such as MPEG-1/2, AAC, AC-3, and etc.

• ETRI Journal
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• v.29 no.1
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• pp.103-106
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• 2007

Pilot-based coding (PBC), which is used for lossless bit rate reduction of audio coding, has been recently proposed for MPEG Surround. We propose extended PBC for further lossless bit rate reduction of MPEG Surround. Extended PBC selects the number of pilots depending on the parameter band number and the type of spatial parameter. It then encodes the pilots and the relevant difference data. Experiments show that extended PBC is more effective than the original PBC, especially for high bit rate modes, with a negligible complexity increase on the decoder side.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.45 no.2
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• pp.81-89
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• 2008

Wideband speech, characterized by a bandwidth of about 7 kHz (50-7000 Hz), provides a substantial quality improvement in terms of naturalness and intelligibility. Although higher data rates are required, it has extended its application to audio and video conferencing, high-quality multimedia communications in mobile links or packet-switched transmissions, and digital AM broadcasting. In this paper, we present a new bandwidth-scalable coder for wideband speech and audio signals. The proposed coder spits 8kHz signal bandwidth into two narrow bands, and different coding schemes are applied to each band. The lower-band signal is coded using the ITU-T G.729/G.729E coder, and the higher-band signal is compressed using a new algorithm based on the gammatone filter bank with an invertible auditory model. Due to the split-band architecture and completely independent coding schemes for each band, the output speech of the decoder can be selected to be a narrowband or wideband according to the channel condition. Subjective tests showed that, for wideband speech and audio signals, the proposed coder at 14.2/18 kbit/s produces superior quality to ITU-T 24 kbit/s G.722.1 with the shorter algorithmic delay.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2018.06a
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• pp.51-52
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• 2018

본 논문에서는 조건부 오토 인코더를 사용하여 오디오 고대역 신호를 부호화 하는 기술을 제안한다. 오토 인코더의 데이터 압축 특성을 이용하여 부호화를 위한 데이터의 양을 크게 줄인다. 제안하는 알고리즘은 기존의 오토 인코더와 달리 과거의 정보가 포함된 2차원 조건을 함께 입력하여 오토 인코더가 코딩 프레임의 고대역을 복원하는 것을 돕도록 한다. 2차원 조건과 입력을 압축하여 연결한 후 디코딩하여 코딩 프레임의 고주파 대역을 만든다. 제안하는 방법을 사용하면 저대역 MDCT 계수와 고대역 MDCT 계수를 오토 인코더로 압축한 결과만으로 원본과 유사한 음질을 청취할 수 있다.

• The Journal of the Acoustical Society of Korea
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• v.27 no.2
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• pp.66-74
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• 2008

In this paper, we propose new techniques that can improve the quality of AAC and SBR encoders comprised in low bitrate HE-AAC. To reduce the pre-echo artifacts often occurring for transient blocks in AAC, we propose an extended Temporal Noise Shaping (sTNS) in which the frequency range is selectively extended down to the low-frequency region. Also, for he high-frequency region being coded by SBR encoder, tones are identified through a sinusoidal modeling and their frequencies are adjusted within the QMF band in order to reduce the noise floor due to aliasing. Spectrograms of the decoded signals were compared and listening tests were conducted to evaluate the proposed algorithm. Results confirmed the effectiveness of the proposed algorithm.