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Featured researches published by Atsushi Murashima.


2000 IEEE Workshop on Speech Coding. Proceedings. Meeting the Challenges of the New Millennium (Cat. No.00EX421) | 2000

A post-processing technique to improve coding quality of CELP under background noise

Atsushi Murashima; Masahiro Serizawa; Kazunori Ozawa

This paper proposes a novel post-processing technique to improve the coding quality of CELP under background noise. It adaptively smoothes both the spectral envelope and the energy of the estimated excitation signal to reduce their temporal fluctuations, which cause the perceptual degradation. The excitation signal is calculated using the synthesized signal and the spectral parameters given from the decoder. Thus, the proposed post-processing is performed separately from the decoder. The smoothing is applied only in non-speech periods and the smoothing strength is controlled depending on the characteristics of the synthesized signal to avoid the degradation in speech and non-stationary noise periods. Subjective test results show that the proposed post-processing improves degradation mean opinion score (DMOS) by 0.2 to 0.4 for noisy speech signals, which are coded by the GSM adaptive multi-rate (AMR) codec.


1999 IEEE Workshop on Speech Coding Proceedings. Model, Coders, and Error Criteria (Cat. No.99EX351) | 1999

Multi-rate wideband speech/channel codec based on MPEG-4/CELP for ETSI/GSM full-rate channel

Atsushi Murashima; M. Serizawa; K. Ozawa

This paper proposes a wideband multi-rate speech and channel codec based on the MPEG-4/CELP for the ETSI/GSM full-rate channel. In order to improve coding performance under mobile environments, such as channel error and background noise, the proposed codec operates at three bit allocations between speech and channel coding with a constant gross bit-rate of 22.8 kbit/s. The speech coding bit-rates are 10.9, 12.1 and 15.9 kbit/s. It achieves high speech quality under any channel condition by switching the bit allocations and also for noisy speech by using the highest bit-rate. The preliminary subjective evaluation tests show the speech quality is improved by switching the bit allocation under error conditions. It is also comparable of superior to ITU-T Recommendation G.722 48 kbit/s for carrier-to-interference ratios (C/I) higher than 10 dB. The codec at 15.9 kbit/s also gives comparable speech quality to G.722 at 48 kbit/s under background noise conditions.


signal processing systems | 2007

Low Complexity Real-Time Video Transcoders for Video Upload and Retrieval Applications

Kazunori Ozawa; Hironori Ito; Kazuteru Watanabe; Kazuhiro Koyama; Atsushi Murashima

This paper proposes low complexity real-time video transcoders for video upload and retrieval applications. The proposed transcoders consist of both picture type conversion and syntax conversion in the compressed domain, and can highly reduce the computation amount in comparison with that of the conventional tandem transcoding method, in which decoding and re-encoding are necessary. The proposed syntax conversion can provide loss-less conversion, while the conventional tandem method is lossy conversion. Timestamp adjustment method is also proposed to avoid audio visual de-synchronization problem due to picture type conversion, when the stream is transmitted via constant bandwidth network. Complexity evaluation results based on DSP implementation confirm that the complexity of the proposed transcoders is very low and is less than 1/3 of that of the conventional tandem transcoder.


Archive | 1999

Apparatus for encoding and apparatus for decoding speech and musical signals

Atsushi Murashima


Archive | 1999

Speech and music signal coder/decoder

Atsushi Murashima; Kazunori Ozawa


Archive | 2000

Speech signal decoding method and apparatus, speech signal encoding/decoding method and apparatus, and program product therefor

Atsushi Murashima


Archive | 2006

Speech signal decoding method and apparatus using decoded information smoothed to produce reconstructed speech signal of enhanced quality

Atsushi Murashima


Archive | 2004

Code conversion method and device

Atsushi Murashima


Archive | 2005

Method and apparatus for transcoding between different speech encoding/decoding systems and recording medium

Atsushi Murashima


Archive | 2002

Code conversion method, apparatus, program, and storage medium

Atsushi Murashima

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