著者

Zhi Zhu Katsuhiko Yamamoto Masashi Unoki Naofumi Aoki

出版者

信号処理学会

雑誌

Journal of Signal Processing (ISSN:13426230)

巻号頁・発行日

vol.18, no.6, pp.303-307, 2014-11-25 (Released:2014-11-25)

参考文献数

7

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Speech scrambling aims to eliminate intelligibility of original speech in order to preventing eavesdropping and copyright infringement. There is, however, a problem in that completely recovering scrambled speech into the original speech cannot be achieved with conventional methods. In this paper, we propose a speech scrambling method that uses the random-bit-shift of quantization bits with common keys. We evaluated the confidentiality and efficiency of the proposed method by using two objective measures, SER and PESQ. As a result we confirmed that speech signals can be scrambled into completely unintelligible sounds with the proposed method. Moreover, it is possible to restore a scrambled speech signal into the original one completely. In addition, we also confirmed that the scrambled speech signal could not be descrambled correctly with the wrong key.

著者

Toshio Irino Masashi Unoki

出版者

Acoustical Society of Japan

雑誌

Journal of the Acoustical Society of Japan (E) (ISSN:03882861)

巻号頁・発行日

vol.20, no.6, pp.397-406, 1999 (Released:2011-02-17)

参考文献数

26

被引用文献数

8 20

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This paper proposes a new auditory filterbank that enables signal resynthesis from dynamic representations produced by a level-dependent auditory filterbank. The filterbank is based on a new IIR implementation of the gammachirp, which has been shown to be an excellent candidate for asymmetric, level-dependent auditory filters. Initially, the gammachirp filter is shown to be decomposed into a combination of a gammatone filter and an asymmetric function. The asymmetric function is excellently simulated with a minimum-phase IIR filter, named the “asymmetric compensation filter”. Then, two filterbank structures are presented each based on the combination of a gammatone filterbank and a bank of asymmetric compensation filters controlled by a signal level estimation mechanism. The inverse filter of the asymmetric compensation filter is always stable because the minimum-phase condition is satisfied. When a bank of inverse filters is utilized after the gammachirp analysis filterbank and the idea of wavelet transform is applied, it is possible to resynthesize signals with small time-invariant errors and achieve a guaranteed precision. This feature has never been accomplished by conventional active auditory filterbanks. The proposed analysis/synthesis gammachirp filterbank is expected to be useful in various applications where human auditory filtering has to be modeled.

著者

Katsuhiko Yamamoto Zhi Zhu Masashi Unoki Naofumi Aoki

出版者

信号処理学会

雑誌

Journal of Signal Processing (ISSN:13426230)

巻号頁・発行日

vol.18, no.4, pp.205-208, 2014-07-30 (Released:2014-07-30)

参考文献数

10

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Speech scrambling methods are widely used for copyright protection and encrypting digital speech signals in order to guarantee the confidentiality of the original signals. They are very important methods for preventing eavesdropping and unauthorized copying. However, it seems to be impossible to completely recover a scrambled speech signal into the original signal. Moreover, nobody can comprehend the partial speech content from speech signals scrambled with these methods. In this paper, we propose a semi-scramble method for speech signals based on phonemic restoration. By using a speech scrambling method based on the random-bit shift of quantization bits, speech signals are converted to scrambled signals in partial intervals. We evaluated the confidentiality and efficiency of the proposed method by using two objective measures, signal-to-error ratio (SER) and perceptual evaluation of speech quality (PESQ). As a result, we confirmed that the proposed method can play a role in copyright protection for an original signal and recover a semi-scrambled speech signal into the original one. Finally, we indicated that the acoustic characteristics of signal semi-scrambled with the proposed method enable the listener to understand the speech information.