检索结果-内蒙古大学图书馆

Hidden data transmission in mixed excitation linear prediction coded speech using quantisation index modulation

IET INFORMATION SECURITY 2010年第3期4卷 158-166页

作者： Yargicoglu, A. U. Ilk, H. G. Aselsan Elect Ind Inc Commun Div TR-06172 Ankara Turkey Ankara Univ Fac Engn Dept Elect Engn TR-06100 Ankara Turkey

In this study novel data hiding methods that embed secret data during mixed excitation linear prediction (MELP) coding of the speech signal are proposed. The secret data bits are hidden by using quantisation index modulation (QIM) which is carried out in the multistage vector quantisation (MSVQ) of line spectral frequencies (LSF) parameters. The distortion rates, introduced by the proposed data hiding methods, are compared with that of a previously published dither-like data hiding method by employing tests performed on the speech files selected from NTIMIT database. Weighted Euclidian distances and spectral distortion are calculated as objective distortion quality metrics, whereas perceptual evaluation of speech quality, PESO P.862, which tries to mimic mean opinion scores, is used to give subjective listening test results. After the calculation of objective and subjective quality metrics, it is determined that the methods with QIM at the third and fourth stages of the MSVQ distort speech only as much as that resulting from a 10(-3) bit error rated (BER) additive white Gaussian noise (AWGN) channel. Later, steganographic strengths of some selected QIM methods are elaborated by the steganalysis methods trained with the computed chaotic features. According to the steganalysis results, methods that apply QIM on the third and fourth indexes of MELP-MSVQ assure both lower distortion and better steganographic imperceptibility.

关键词： steganalysis methods data hiding methods multistage vector quantisation hidden data transmission spectral distortion Cryptography AWGN channels quantisation index modulation line spectral frequency parameter steganographic strengths mean opinion scores PESQ P.862 speech quality mixed excitation linear prediction coding steganography speech processing techniques speech coding weighted Euclidian distances speech and audio coding Data security linear prediction coded speech additive white Gaussian noise vector quantisation

来源：评论

学校读者我要写书评

暂无评论

Autoregressive Models of Amplitude Modulations in audio Compression

引用

IEEE TRANSACTIONS ON audio speech AND LANGUAGE PROCESSING 2010年第6期18卷 1624-1631页

作者： Ganapathy, Sriram Motlicek, Petr Hermansky, Hynek Johns Hopkins Univ Elect & Comp Engn Dept Baltimore MD 21218 USA Idiap Res Inst CH-1920 Martigny Switzerland

We present a scalable medium bit-rate wide-band audio coding technique based on frequency-domain linear prediction (FDLP). FDLP is an efficient method for representing the long-term amplitude modulations of speech/audio signals using autoregressive models. For the proposed audio codec, relatively long temporal segments (1000 ms) of the input audio signal are decomposed into a set of critically sampled sub-bands using a quadrature mirror filter (QMF) bank. The technique of FDLP is applied on each sub-band to model the sub-band temporal envelopes. The residual of the linear prediction, which represents the frequency modulations in the sub-band signal, are encoded and transmitted along with the envelope parameters. These steps are reversed at the decoder to reconstruct the signal. The proposed codec utilizes a simple signal independent nonadaptive compression mechanism for a wide class of speech and audio signals. The subjective and objective quality evaluations show that the reconstruction signal quality for the proposed FDLP codec compares well with the state-of-the-art audio codecs in the 32-64 kbps range.

关键词： frequency-domain linear prediction (FDLP) modulation spectrum objective and subjective evaluation of audio quality speech and audio coding

来源：评论

学校读者我要写书评

暂无评论

Switched adaptive quantiser for speech compression based on optimal companding and correlation

引用

IET SIGNAL PROCESSING 2011年第7期5卷 701-707页

作者： Despotovic, V. Peric, Z. Velimirovic, L. Delic, V. Univ Belgrade Tech Fac Bor Bor 19210 Serbia Univ Nis Fac Elect Engn Nish 18000 Serbia Univ Novi Sad Fac Tech Sci Novi Sad 21000 Serbia

This study describes a novel adaptive quantiser based on the optimal companding technique. Adaptation is achieved by adjusting the input of the fixed or non-adaptive quantiser according to the estimated and quantised gain on each particular frame. In such a way better quantiser adaptation to the varying input statistics is provided. Selection of the appropriate bit rate is performed depending on the value of the correlation coefficient rho on each frame. The decision thresholds for rho are determined under the condition that the signal to quantisation noise ratio does not drop under 34.3 dB, satisfying the G. 712 standard quality of speech, while decreasing the bit rate. The information about the gain and about the chosen bit rate is then transferred as a side information to a decoder. Although this slightly increases the side information, the overall savings in the bit rate have shown to be substantial. Theoretical and experimental results are provided, which point out the benefits that can be achieved using the proposed algorithm.

关键词： data compression speech coding speech standard quality signal to quantisation noise ratio switched nonadaptive quantiser optimal companding technique speech processing techniques noise figure 34.3 dB speech and audio coding optimal correlation speech compression

来源：评论

学校读者我要写书评

暂无评论

Voice phishing detection technique based on minimum classification error method incorporating codec parameters

引用

IET SIGNAL PROCESSING 2010年第5期4卷 502-509页

作者： Chang, J. -H. Lee, K. -H. Inha Univ Sch Elect Engn Inchon 402751 South Korea

The authors propose an effective voice phishing detection algorithm based on a Gaussian mixture model (GMM) employing the minimum classification error (MCE) technique. The detection of voice phishing is performed based on the GMM using decoding parameters of the 3GPP2 selectable mode vocoder (SMV) codec directly extracted from the decoding process of the transmitted speech information in the mobile phone. The authors' approach is further improved by the MCE scheme in that different weights are assigned to each likelihood ratio and is considered to be new. The experimental results show that the proposed method is effective in discriminating between true statements and lies.

关键词： decoding process Data security telecommunication security mobile phone Mobile radio systems codec parameter Other topics in statistics mobile handsets vocoders decoding minimum classification error method GMM speech coding speech information transmission 3G mobile communication speech and audio coding Gaussian processes 3GPP2 selectable mode vocoder Gaussian mixture model speech processing techniques voice phishing detection

来源：评论

学校读者我要写书评

暂无评论

Objective quality measures for perceptual evaluation in digital audio watermarking

引用

IET SIGNAL PROCESSING 2011年第7期5卷 623-631页

作者： Lin, Y. Abdulla, W. Univ Auckland Dept Elect & Comp Engn Auckland 1142 New Zealand

In this study, objective quality measures adopted in speech processing are proposed to evaluate the perceptual quality of the watermarked audio signals. Under different audio watermarking techniques such as spread spectrum watermarking, cepstrum domain watermarking, wavelet domain watermarking, echo hiding and histogram-based watermarking, the dissimilarities between the watermarked and host audio signals are separately quantified in terms of several well-developed quality measures, including the signal-to-noise ratio (SNR), the segmental SNR (segSNR), the cepstral distortion (CD), the log-likelihood ratio (LLR), the Itakura-Saito (IS) distortion, the log-area ratio (LAR) and the weighted spectral slope (WSS) measures. For correlation analysis, subjective listening tests and a commercial evaluation tool 'Perceptual Model-Quality Assessment' (PEMO-Q) are also used to grade the differences. Two types of Pearson's correlation coefficients, that is, average and overall correlation coefficients, are calculated to evaluate the performance of quality measures serving as the predictors of perceptual quality. Extensive experiments indicate that there is a good correlation between the investigated quality measures and subjective difference grades. Specifically, the LAR and LLR measures yield the highest average and overall correlation coefficients for most audio watermarking techniques. Moreover, a large decrease in computation time reveals that quality measures run much faster than PEMO-Q. It is therefore concluded that quality measures can be used for objective assessment of the perceptual quality in audio watermarking.

关键词： cepstral distortion Data security Itakura-Saito distortion log-area ratio signal-to-noise ratio log-likelihood ratio wavelet domain watermarking correlation methods audio watermarking spread spectrum watermarking cepstrum domain watermarking correlation analysis Pearson-s correlation coefficients speech recognition and synthesis equipment objective quality measures segmental SNR histogram-based watermarking speech and audio coding digital audio watermarking perceptual evaluation speech processing techniques weighted spectral slope measures echo hiding speech processing

来源：评论

学校读者我要写书评

暂无评论

Robust SBR method for adverse Mandarin speech recognition

引用

ELECTRONICS LETTERS 1999年第11期35卷 875-876页

作者： Hong, WT Chen, SH Natl Chiao Tung Univ Dept Commun Engn Hsinchu Taiwan

An RNN-based robust signal bias removal (RRSBR) method is proposed for improving both the recognition performance and the computational efficiency of the SBR method fbr adverse Mandarin speech recognition. It differs from the SBR method in using three broad-class sub-codebooks to encode the feature vector of each frame and combining the three encoding residuals to form the frame-level signal bias estimate. A novel approach involving softly combining the board-class encoding residuals using dynamic weighting functions generated by an RNN is applied. Experimental results show that the RRSBR method significantly outperforms the SBR method.

关键词： adverse Mandarin speech recognition recurrent neural network Neural nets recurrent neural nets speech coding RNN-based robust signal bias removal method speech recognition recognition performance feature vector encoding dynamic weighting functions speech processing techniques speech recognition frame-level signal bias estimate robust SBR method computational efficiency Neural computing techniques speech and audio coding board-class encoding residuals speech recognition and synthesis broad-class sub-codebooks

来源：评论

学校读者我要写书评

暂无评论

Multiple length adaptive windowing of audio signals

引用

ELECTRONICS LETTERS 1999年第14期35卷 1133-1135页

作者： Vanelli-Coralli, A Corazza, GE Orsini, A Univ Bologna Dept Elect Comp Sci & Syst I-40136 Bologna Italy

audio coders generally operate on a per-frame basis to tackle signal non-stationarity. The choice of frame length is crucial to system performance since it trades off coding efficiency and pre-echo avoidance. A new algorithm based on multiple length adaptive windowing is proposed as an enhanced front-end for audio coders. Overhead reductions in excess of 40% are reported.

关键词： audio signals coding efficiency audio coding frame length multiple length adaptive windowing overhead reductions enhanced front-end pre-echo avoidance audio coders speech and audio coding system performance adaptive signal processing signal nonstationarity

来源：评论

学校读者我要写书评

暂无评论

Accurate, critically sampled characteristic waveform surface construction for waveform interpolation decomposition

引用

ELECTRONICS LETTERS 2000年第14期36卷 1245-1247页

作者： Chong-White, NR Burnett, IS Univ Wollongong Whisper Labs Wollongong NSW 2522 Australia

A waveform-matched waveform interpolation (WMWI) technique is presented which offers improved signal analysis over standard WI coders and, in the unquantised case, perfect reconstruction. In WMWI, an accurate representation of speech evolution is formed by extracting consecutive pitch periods of a time-warped, constant pitch residual. A pitch track optimisation technique is described which ensures that the critically sampled pitch periods can be effectively decomposed into a slowly evolving and rapidly evolving waveform, allowing efficient quantisation.

关键词： waveform interpolation decomposition pitch track optimisation technique quantisation (signal) speech coding consecutive pitch periods extraction Interpolation and function approximation (numerical analysis) Signal processing theory characteristic waveform surface construction quantisation signal reconstruction optimisation time-warped constant pitch residual perfect reconstruction signal analysis critically sampled pitch periods waveform-matched waveform interpolation technique interpolation Optimisation techniques speech and audio coding

来源：评论

学校读者我要写书评

暂无评论

Efficient quantisation method for LSF parameters based on restricted temporal decomposition

引用

ELECTRONICS LETTERS 1999年第12期35卷 962-964页

作者： Kim, SJ Oh, YH Korea Adv Inst Sci & Technol Dept Comp Sci Taejon 305701 South Korea

A restricted temporal decomposition method is presented for line spectral frequency (LSF) parameters. The proposed method interpolates an LSF vector trajectory efficiently while consenting the LSF ordering property. E... 详细信息

关键词： speech processing techniques interpolation LSF ordering property conservation quantisation method LSF vector trajectory vector quantisation restricted temporal decomposition VQ LSF parameters speech coding line spectral frequency parameters speech and audio coding Interpolation and function approximation (numerical analysis) speech processing Signal processing theory

来源：评论

学校读者我要写书评

暂无评论

Unequal error protection and source-channel decoding of CELP speech

引用

ELECTRONICS LETTERS 2002年第7期38卷 347-349页

作者： Nazer, A Alajaji, F Queens Univ Dept Math & Stat Kingston ON K7L 3N6 Canada

The reliable communication of FS CELP 10 16 encoded speech over very noisy channels is investigated. Using second-order Markov chains it is shown that over one-quarter of the CELP bits in every frame of speech are redundant. An unequal error protection coding scheme, which exploits this residual redundancy, is proposed for sending the CELP parameters over Gaussian and Rayleigh fading channels. Simulations indicate substantial coding gains over conventional systems.

关键词： speech coding Gaussian channels combined source-channel coding reliable communication coding gains error correction codes voice communication decoding CELP speech second-order Markov chains linear predictive coding source-channel decoding Rayleigh channels telecommunication network reliability Markov processes unequal error protection coding FS CELP 1016 encoded speech redundancy residual redundancy very noisy channels speech and audio coding Rayleigh fading channels

来源：评论

学校读者我要写书评

暂无评论

建议与咨询留下您的常用邮箱和电话号码，以便我们向您反馈解决方案和替代方法

时间限定

文献类型

馆藏选择

核心期刊

语言

文献类型

帮助

文字说明：

检索规则说明：

检索范例：

分类表

所选分类

限定检索结果

文献类型

馆藏范围

日期分布

学科分类号

主题

机构

作者

语言

请选择保存的检索档案：

请选择收藏分类：

通借通还

建议与咨询 留下您的常用邮箱和电话号码，以便我们向您反馈解决方案和替代方法

时间限定

文献类型

馆藏选择

核心期刊

语言

文献类型

帮助

文字说明：

检索规则说明：

检索范例：

分类表

所选分类

限定检索结果

文献类型

馆藏范围

日期分布

学科分类号

主题

机构

作者

语言

请选择保存的检索档案： 新增检索档案 确定 取消

请选择收藏分类： 新增自定义分类 确定 取消

通借通还

建议与咨询留下您的常用邮箱和电话号码，以便我们向您反馈解决方案和替代方法

请选择保存的检索档案：

请选择收藏分类：