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Dive into the research topics where Yuichiro Takamizawa is active.

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Featured researches published by Yuichiro Takamizawa.


Journal of the Acoustical Society of America | 2006

Audio decoding apparatus and method for band expansion with aliasing adjustment

Naoya Tanaka; Osamu Shimada; Mineo Katano-shi Tsushima; Takeshi Kobe-shi Norimatsu; Kok Seng Chong; Kim Hann Kuah; Sua Hong Neo; Toshiyuki Nomura; Yuichiro Takamizawa; Masahiro Serizawa

An audio decoding apparatus decodes high frequency component signals using a band expander that generates multiple high frequency subband signals from low frequency subband signals divided into multiple subbands and transmitted high frequency encoded information. The apparatus is provided with an aliasing detector and an aliasing remover. The aliasing detector detects the degree of occurrence of aliasing components in the multiple high frequency subband signals generated by the band expander. The aliasing remover suppresses aliasing components in the high frequency subband signals by adjusting the gain used to generate the high frequency subband signals. Thus occurrence of aliasing can be suppressed and the resulting degradation in sound quality can be reduced, even when real-valued subband signals are used in order to reduce the number of operations.


international conference on acoustics, speech, and signal processing | 2001

High-quality and processor-efficient implementation of an MPEG-2 AAC encoder

Yuichiro Takamizawa; Toshiyuki Nomura; Masao Ikekawa

Presents MPEG-2 AAC LC Profile encoder software for an Intel Pentium III processor. Modified discrete cosine transform (MDCT) and quantization processing are accelerated by the use of SIMD instructions. Psycho-acoustic analysis in the MDCT domain makes the use of FFTs unnecessary. Better sound quality is provided by greater efficiency in quantization processing and Huffman coding. All of this results in high-quality and processor-efficient implementation of an MPEG-2 AAC encoder. Sound quality achieved at 96 kbps/stereo is significantly better than that of MP3 at the same bitrate. The encoder works 13 times faster than realtime for stereo encoding on an 800MHz Pentium III processor.


signal processing systems | 1999

MPEG-2 AAC 5.1-channel decoder software for a low-power embedded RISC microprocessor

Yuichiro Takamizawa; Kouhei Nadehara; Max Boegli; Masao Ikekawa; Ichiro Kuroda

Presented here is MPEG-2 AAC low complexity profile decoder software for a low-power embedded RISC microprocessor, NEC V830 (300 mW @133 MHz). Fast processing methods for IMDCT reduce execution time by 41% and help achieve real-time decoding of a 5.1-channel audio signal, while using only 64.7% of processor capacity.


pacific rim conference on multimedia | 2003

Low power spectral band replication technology for the MPEG-4 audio standard

Chong Kok Seng; Naoya Tanaka; Toshiyuki Nomura; Osamu Shimada; Kuah Kim Hann; Mineo Tsushima; Yuichiro Takamizawa; Neo Sua Hong; Takeshi Norimatsu; Masahiro Serizawa

Spectral band replication (SBR) is the bandwidth extension technology for the MPEG-4 audio extension 1 standard. Its principle is based on mapping the low-frequency portion of an audio signal coded at low bitrate to the missing high-frequency region, and uses a small amount of information embedded in the audio bitstream to shape the energy envelop and tone to a noise ratio of the mapped signals, such that they aurally resemble the high-frequency spectrum of the original signal, and thereby deliver compact disc-like listening sensation to the listeners. Low power SBR (LP-SBR) is a simplified version of the SBR technology that applies real-valued processing to all SBR modules, and applies aliasing reduction tools to suppress the resultant aliasing artifacts. LP-SBR requires 40% less computational cost compared to the original SBR. This paper describes the causes of aliasing artifacts and the principles behind the anti-aliasing solutions.


international conference on acoustics, speech, and signal processing | 1997

An efficient tonal component coding algorithm for MPEG-2 Audio NBC

Yuichiro Takamizawa; Masahiro Iwadare; Akihiko Sugiyama

This paper proposes a tonal component coding algorithm for a codec that employs a transform followed by Huffman coding, such as MPEG-2 Audio NBC (non-backward compatible). After the input audio signal is mapped onto a frequency domain, the proposed algorithm withdraws local maximum components that degrade coding efficiency. By this withdrawal, the flatness of the spectrum increases and the efficiency in Huffman coding is improved. The withdrawn components are encoded separately as side information. When the frequency resolution of the time/frequency mapping is high, this algorithm works more effectively since local maximum samples appear more frequently with such a mapping. Simulation results show that this algorithm achieves as much as 11% bit reduction per frame and improves the coding efficiency in 41% of all the audio frames.


Archive | 2003

Audio decoding apparatus and method

Naoya Tanaka; Osamu Shimada; Mineo Tsushima; Takeshi Norimatsu; Kok Seng Chong; Kim Hann Kuah; Sua Hong Neo; Toshiyuki Nomura; Yuichiro Takamizawa; Masahiro Serizawa


Archive | 2002

Device and method for audio decoding, and program

Toshiyuki Nomura; Takeshi Norimatsu; Masahiro Serizawa; Osamu Shimada; Yuichiro Takamizawa; Naoya Tanaka; Mineo Tsushima; 則松 武志; 嶋田 修; 津島 峰生; 田中 直也; 芹沢 昌宏; 野村 俊之; 高見沢 雄一郎


Archive | 2003

Audio decoding apparatus and audio decoding method

Mineo Tsushima; Naoya Tanaka; Takeshi Norimatsu; Kok Seng Chong; Kim Hann Kuah; Sua Hong Neo; Toshiyuki Nomura; Osamu Shimada; Yuichiro Takamizawa; Masahiro Serizawa


Archive | 2002

Speech band expansion system and method therefor

Toshiyuki Nomura; Yuichiro Takamizawa; 俊之 野村; 雄一郎 高見沢


Archive | 2001

Audio encoder and psychoacoustic analyzing method therefor

Satoshi Hasegawa; Yuichiro Takamizawa

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