• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.5
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• pp.89-99
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• 2010

A unified method for separating musical sources, singing voice for example, from stereophonic mixtures is provided. We usually have two observed signals in stereophonic music contents, where more than two instruments are played together. If we regard each instrument as source, this problem becomes an underdetermined source separation problem and cannot be solved by conventional methods, which infers the spatial environment of the downmixing process happens. Instead, source-specific information has been exploited to recover a particular instrumental source. This paper provides a unifying structure consists of heterogenious ad-hoc separate algorithms, which are designed for separating vocal sources using stereophonic channel information and dominant pitch information of the sources, respectively. Experiments on real world music contents show that the proposed unification can neutralize the drawbacks of the two ad-hoc separation algorithms and finally enhance the separation results.

• Journal of Intelligence and Information Systems
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• v.24 no.2
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• pp.21-35
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• 2018

Since the introduction of MP3 players, CD recordings have gradually been vanishing, and the music consuming environment of music users is shifting to mobile devices. The introduction of smart devices has increased the utilization of music through music playback, mass storage, and search functions that are integrated into smartphones and tablets. At the time of initial MP3 player supply, the bitrate of the compressed music contents generally was 128 Kbps. However, as increasing of the demand for high quality music, sound quality of 384 Kbps appeared. Recently, music content of FLAC (Free License Audio Codec) format using lossless compression method is becoming popular. The download service of many music sites in Korea has classified by unlimited download with technical protection and limited download without technical protection. Digital Rights Management (DRM) technology is used as a technical protection measure for unlimited download, but it can only be used with authenticated devices that have DRM installed. Even if music purchased by the user, it cannot be used by other devices. On the contrary, in the case of music that is limited in quantity but not technically protected, there is no way to enforce anyone who distributes it, and in the case of high quality music such as FLAC, the loss is greater. In this paper, the author proposes an audio watermarking technology for copyright protection of high quality stereo music. Two kinds of information, "Copyright" and "Copy_free", are generated by using the turbo code. The two watermarks are composed of 9 bytes (72 bits). If turbo code is applied for error correction, the amount of information to be inserted as 222 bits increases. The 222-bit watermark was expanded to 1024 bits to be robust against additional errors and finally used as a watermark to insert into stereo music. Turbo code is a way to recover raw data if the damaged amount is less than 15% even if part of the code is damaged due to attack of watermarked content. It can be extended to 1024 bits or it can find 222 bits from some damaged contents by increasing the probability, the watermark itself has made it more resistant to attack. The proposed algorithm uses quantization in DCT so that watermark can be detected efficiently and SNR can be improved when stereo music is converted into mono. As a result, on average SNR exceeded 40dB, resulting in sound quality improvements of over 10dB over traditional quantization methods. This is a very significant result because it means relatively 10 times improvement in sound quality. In addition, the sample length required for extracting the watermark can be extracted sufficiently if the length is shorter than 1 second, and the watermark can be completely extracted from music samples of less than one second in all of the MP3 compression having a bit rate of 128 Kbps. The conventional quantization method can extract the watermark with a length of only 1/10 compared to the case where the sampling of the 10-second length largely fails to extract the watermark. In this study, since the length of the watermark embedded into music is 72 bits, it provides sufficient capacity to embed necessary information for music. It is enough bits to identify the music distributed all over the world. 272 can identify $4*10^{21}$, so it can be used as an identifier and it can be used for copyright protection of high quality music service. The proposed algorithm can be used not only for high quality audio but also for development of watermarking algorithm in multimedia such as UHD (Ultra High Definition) TV and high-resolution image. In addition, with the development of digital devices, users are demanding high quality music in the music industry, and artificial intelligence assistant is coming along with high quality music and streaming service. The results of this study can be used to protect the rights of copyright holders in these industries.

• Broadcasting and Media Magazine
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• v.16 no.4
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• pp.59-71
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• 2011

오디오와 음성 압축 기술적 근간은 서로 다르지만, 최근의 모바일 멀티미디어 기기 시장의 컨버전스 현상에 따라 압축하고자 하는 신호가 혼용되고 있으며, 비슷한 목표 전송률과 음질로 수렴하고 있다. 현재는 동일 기기에서 서로 다른 압축 기술을 적용하고 있으나, 음성과 음악이 동시에 서비스 되는 멀티미디어 기기에서는 단일 압축 방식으로 처리하고자 하는 이슈가 부각되고 있다. 특히, 스마트 폰 및 음악 콘텐츠 포탈 서비스의 대중화를 고려할 때, 음성 및 음악 신호 모두를 효율적으로 압축하는 음악 및 음성 신호의 융합 압축 기술이 더욱 필요해 보인다. 본 고에서는 MPEG 오디오 그룹에서 가장 최근 진행한 Unified Speech and Audio Coding(USAC)의 탄생 배경 및 표준화 현황을 소개한다. USAC는 64kbps 이하에서 기술적으로 최고 성능을 지닌 AMR-WB+ 및 HE-AAC v2보다도 우월한 음질을 보이며, 높은 비트율에서도 동등한 음질을 보장한다. 이런 우수한 음질에 기여한 USAC의 스위칭 구조와 더불어 기술적으로 향상된 주요 모듈인 파라미터 기반 스테레오 및 고주파 압축, 그리고 엔트로피 코딩 방식에 대해서 살펴 본다. 향후, 다양한 오디오 신호를 효율적으로 압축하는 USAC는 디지털 라디오, 모바일 TV, 그리고 오디오 북과 같은 사용자 시나리오에서 사용될 확률이 높아 보인다. 또한, USAC는 배경 잡음이나 배경 음악이 있는 경우에도 성능이 우수하기 때문에 YouTube 및 podcast 등과 같이 사용자가 콘텐츠를 생성할 때도 유용하게 사용 될 수 있다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2010.10a
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• pp.268-270
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• 2010

Recently, According to increasing interest to original sound Karaoke instrument, MIDI type karaoke manufacturer attempt to make more cheap method instead of original recoding method. In this paper, we developed how to create MR from AR, recorded in stereo, by using the energy difference in the frequency domain and how to implement in DSP(TMS320C6713) were developed. At the output of the DSP board, 6-channel audio output interface designed for real-time stereophonic generating original sound, vocals removed MR, and separated vocals simultaneously. Real-time listening test using DSP show vocal separating and removal task successfully.

• The Journal of the Korea institute of electronic communication sciences
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• v.10 no.1
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• pp.1-6
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• 2015

Recently, According to increasing interest to original sound Karaoke instrument, MIDI type karaoke manufacturer attempt to make more cheap method instead of original recoding method. The specific method is to make the original sound accompaniment to remove only the voice of the singer in the singer music album. In this paper, a system to separate vocal components from music accompaniment for stereo recordings were proposed. Proposed system consists of two stages. The first stage is a vocal detection. This stage classifies an input into vocal and non vocal portions by using SVM with MFCC. In the second stage, selective frequency subtractions were performed at each frequency bin in vocal portions. Listening test with removed vocal music from proposed system show relatively high satisfactory level.

• Journal of Broadcast Engineering
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• v.20 no.3
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• pp.408-413
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• 2015

Recently, According to increasing interest to original sound Karaoke instrument, MIDI type karaoke manufacturer attempt to make more cheap method instead of original recoding method. The specific method is to make the original sound accompaniment to remove only the voice of the singer in the singer music album. In this paper, a system to separate vocal components from music accompaniment for stereo recordings were proposed. Proposed system consists of two stages. The first stage is a vocal detection. This stage classifies an input into vocal and non vocal portions by using SVM with MFCC. In the second stage, selective frequency subtractions were performed at each frequency bin in vocal portions. In this case, it is determined in consideration not only the energies for each frequency bin but also the phase of the each frequency bin at each channel signal. Listening test with removed vocal music from proposed system show relatively high satisfactory level.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.427-431
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• 1998

스테레오, MIDI, MP3 등등의 신호를 입력으로 하여 음상의 공간감, 거리감, 방향감, 확산감, 위치감 등이 지각을 느끼게 하는 공간적 현장감 시스템을 제안한다. 본 제안은 2채널로 기록된 매체를 2채널 또는 다채널로 표현이 가능하며 영상과 더불어 두 개의 스피커만으로도 입체 음향을 즐길 수 있다. 음악의 경우는 장르에 따라 각각의 특징을 가지고 있다. 그러한 특징은 위치감과 방향감에 의해 음장의 형태를 구현하고 공간감 거리감을 부과하므로 입체 음장 구현을 도모한다. 그리고 확산감을 부과하므로써 실조화 공간 음장을 구현할 수있다. 본 논문은 특정한 음색을 변화하기 위하여 이퀄라이저를 이용할 필요가 없으며, 다양한 음장 형태를 DSP를 이용 알고리즘화 하여 구현하던 것을 본 RSF는 아날로그 방식으로 구현하므로써 노이즈 측면과 PCB 패턴 부분 고려등을 고려하지 않고도 구현할 수 있다.

• Journal of the Institute of Electronics and Information Engineers
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• v.53 no.11
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• pp.82-88
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• 2016

In this paper, we propose an intelligibility enhancement algorithm for multimedia contents using spectral shaping. The dialogue signals is essential to understand the plot of audio-visual media contents such as movie and TV. However, the non-dialogue components as like sound effects and background music often degrade the dialogue clarity. To overcome this problem, this paper tries to improves the dialogue clarity of audio soundtracks which contain important cues for the visual scenes. In the proposed method, the dialogue components are first detected by soft masker based on speech presence probability (SPP) which is widely used in speech enhancement field. Then, extracted dialogue signals are applied to the spectral shaping method. It reallocate the spectral-temporal energy of speech to enhanced the intelligibility. The total energy is maintained as unchanged via a loudness normalization process to prevent saturation. The algorithm was evaluated using the modeled and real movie soundtracks and it was shown that the proposed algorithm enhances the dialogue clarity while preserving the total audio power.

• Journal of Broadcast Engineering
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• v.16 no.5
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• pp.693-704
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• 2011

As mobile devices become multi-functional, and converge into a single platform, there is a strong need for a codec that is able to provide consistent quality for speech and music contents. MPEG-D USAC standardization activities started at the 82nd MPEG meeting with a CfP and approved Study on DIS at the 96th MPEG meeting. MPEG-D USAC is converged technology of AMR-WB+ and HE-AAC V2. Specifically, USAC utilizes three core codecs (AAC, ACELP, and TCX) for low frequency regions, SBR for high frequency regions, the MPEG Surround for stereo information, and window transition technology for smoothing transition between various core coder. USAC can provide consistent sound quality for both speech and music contents and can be applied to various applications such as multi-media download to mobile devices, digital radio, mobile TV and audio books.

• Journal of Broadcast Engineering
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• v.19 no.3
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• pp.296-306
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• 2014

This paper addresses a problem of non-dialog section detection for the DVS authoring, the goal of which is to find meaningful section from the broadcasting audio, where audio description can be inserted. The broadcasting audio involves the presence of various sounds so that it first discriminates between speech and non-speech for each audio frame. Proposed method jointly exploits the inter-channels structure and speech source characteristics of the broadcasting audio whose number of channel is stereo. Also, rule based post-processing is finally applied to detect the non-dialog section whose length is appropriate for audio description. Proposed method provides more accurate detection compared to conventional method. Experimental results on real broadcasting contents show that qualitative superiority of the proposed method.