检索结果-内蒙古大学图书馆

您好，读者！请登录

内蒙古大学图书馆

首页
概况
党建
资源
服务
科研支持
- 论文收录引用证明
- 科技查新
知识产权
档案馆
帮助

咨询与建议

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

您的常用邮箱：*

您的手机号码：*

问题描述：

当前已输入0个字，您还可以输入200个字

全部搜索
期刊论文
图书
学位论文
标准
纸本馆藏
外文资源发现
数据库导航
超星发现

高级检索

时间限定

出版年份：

文献类型

图书期刊文献学位论文多媒体

馆藏选择

电子馆藏纸本馆藏

核心期刊

全部期刊 SCI 收录期刊 SSCI 收录期刊 EI 收录期刊 CSCD 收录期刊 CSSCI 收录期刊

语言

中文英文

文献类型

期刊文献图书学位论文标准纸本馆藏

帮助

文字说明：

T=题名（书名、题名），A=作者（责任者），K=主题词，P=出版物名称，PU=出版社名称，O=机构（作者单位、学位授予单位、专利申请人），L=中图分类号，C=学科分类号，U=全部字段，Y=年（出版发行年、学位年度、标准发布年）

检索规则说明：

AND代表“并且”；OR代表“或者”；NOT代表“不包含”；(注意必须大写,运算符两边需空一格)

检索范例：

范例一：(K=图书馆学 OR K=情报学) AND A=范并思 AND Y=1982-2016
范例二：P=计算机应用与软件 AND (U=C++ OR U=Basic) NOT K=Visual AND Y=2011-2016

分类表

所选分类

>> <<

限定检索结果

文献类型

2,046 篇 会议
427 篇 期刊文献
1 册 图书

馆藏范围

2,474 篇 电子文献
0 种 纸本馆藏

日期分布

学科分类号

622 篇 工学
- 468 篇 电气工程
- 136 篇 计算机科学与技术...
- 113 篇 信息与通信工程
- 82 篇 电子科学与技术（可...
- 25 篇 生物医学工程（可授...
- 24 篇 软件工程
- 22 篇 控制科学与工程
- 19 篇 仪器科学与技术
- 6 篇 机械工程
- 5 篇 材料科学与工程（可...
- 4 篇 测绘科学与技术
- 3 篇 石油与天然气工程
- 3 篇 船舶与海洋工程
- 3 篇 网络空间安全
- 2 篇 力学（可授工学、理...
- 2 篇 光学工程
- 2 篇 交通运输工程
241 篇 理学
- 208 篇 物理学
- 16 篇 系统科学
- 14 篇 生物学
- 5 篇 数学
- 3 篇 化学
- 3 篇 地球物理学
- 2 篇 统计学（可授理学、...
51 篇 医学
- 42 篇 临床医学
- 11 篇 特种医学
- 4 篇 基础医学(可授医学...
14 篇 管理学
- 14 篇 管理科学与工程(可...
9 篇 文学
- 4 篇 外国语言文学
- 2 篇 新闻传播学
3 篇 农学
3 篇 军事学
2 篇 艺术学
1 篇 教育学

主题

2,474 篇 linear predictiv...
779 篇 speech coding
706 篇 speech analysis
550 篇 speech synthesis
412 篇 bit rate
410 篇 speech recogniti...
350 篇 speech processin...
329 篇 filters
281 篇 frequency
270 篇 testing
254 篇 vocoders
241 篇 vector quantizat...
238 篇 speech enhanceme...
227 篇 quantization
215 篇 predictive model...
202 篇 signal processin...
184 篇 cepstral analysi...
183 篇 speech
179 篇 signal synthesis
178 篇 signal analysis

机构

32 篇 school of electr...
24 篇 at and t bell la...
22 篇 bolt beranek and...
19 篇 bell laboratorie...
11 篇 at and t bell la...
10 篇 naval research l...
10 篇 department of el...
10 篇 ntt human interf...
10 篇 lpcs enserg gren...
9 篇 lpcs-enserg gren...
8 篇 centre for commu...
8 篇 signal technolog...
8 篇 department of el...
8 篇 cselt torino
8 篇 acoustics resear...
7 篇 electrical engin...
7 篇 acoustics resear...
7 篇 comsat laborator...
7 篇 central research...
7 篇 inrs-télécommuni...

作者

19 篇 s. cristoloveanu
18 篇 l. rabiner
14 篇 a.m. kondoz
14 篇 j. makhoul
13 篇 t. barnwell
13 篇 p. mermelstein
12 篇 t. moriya
12 篇 b. atal
12 篇 a. gersho
12 篇 p. hedelin
11 篇 g. ghibaudo
11 篇 gray rm
10 篇 a. rosenberg
10 篇 rabiner lr
10 篇 k. ozawa
9 篇 t.p. barnwell
9 篇 p. kabal
9 篇 a. mccree
9 篇 b.g. evans
8 篇 v. viswanathan

语言

2,290 篇 英文
176 篇 其他
4 篇 中文
3 篇 土耳其文

检索条件"主题词=Linear predictive coding"

共 2474 条记录，以下是1011-1020 订阅

全选清除本页清除全部题录导出标记到"检索档案"

详细简洁

排序：

A low-power DSP core architecture for low bitrate speech codec

A low-power DSP core architecture for low bitrate speech cod...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： H. Okuhata M.H. Miki T. Onoye I. Shirakawa Department of Information Systems Engineering Osaka University Suita Osaka Japan

A VLSI implementation of a low-power DSP is described, which is dedicated to the G.723.1 low bitrate speech codec. A number of sophisticated DSP microarchitectures are devised mainly on dual multiply accumulators, rounding and saturation mechanisms, and two-banked on-chip memory. The proposed DSP architecture has been integrated in a total area of 7.75 mm/sup 2/ by using a 0.35 /spl mu/m CMOS technology, which can operate at 10 MHz with the dissipation of 45 mW from a single 3 V supply.

关键词： Digital signal processing Bit rate Speech codecs Decoding Clocks Encoding Filters CMOS technology Frequency linear predictive coding

来源：评论

学校读者我要写书评

暂无评论

A robust algorithm for formant frequency extraction of noisy speech

A robust algorithm for formant frequency extraction of noisy...

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： Qifang Zhao T. Shimamura J. Suzuki Department of Information and Computer Sciences Saitama University Urawa Saitama Japan

In this paper a new method for formant frequency estimation of noisy speech is proposed based on linear prediction analysis. Usually the linear prediction analysis based algorithms can extract the formant frequencies effectively for clean speech. When speech is corrupted by noise, however, their performance degrades seriously. It is well known that the autocorrelation function has the property of concentrating the energy of the white noise in the vicinity of the zero lag. Utilizing this property of the autocorrelation function, the proposed method extracts the formant frequencies from the autocorrelation function of the speech instead of the speech itself. The experimental results show that the proposed method is much more robust to noise than the conventional linear prediction based algorithms.

关键词： Speech analysis Frequency estimation White noise linear predictive coding Autocorrelation Noise robustness Data mining Speech enhancement Speech processing Algorithm design and analysis

来源：评论

学校读者我要写书评

暂无评论

A 2.4 kbps variable bit rate ADP-CELP speech coder

A 2.4 kbps variable bit rate ADP-CELP speech coder

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： M. Oshikiri M. Akamine Kansai Research Laboratories Toshiba Corporation Kobe Japan

This paper presents a variable bit rate ADP-CELP (adaptive density pulse code excited linear prediction) coder that selects one of four kinds of coding structure in each frame based on short time speech characteristics. To improve speech quality and reduce the average bit rate, we have developed a speech/non-speech classification method using spectrum envelope variation, which is robust for background noise. In addition, we propose an efficient pitch lag coding technique. The technique interpolates consecutive frame pitch lags and quantizes a vector of relative pitch lags consisting of variation between an estimated pitch lag and a target pitch lag in plural subframes. The average bit rate of the proposed coder was approximately 2.4 kbps for speech sources with activity factor of 60%. Our subjective testing indicates the quality of the proposed coder exceeds that of the Japanese digital cellular standard with rate of 3.45 kbps.

关键词： Bit rate Speech coding Working environment noise Testing linear predictive coding Laboratories Robustness Code standards Entropy Oral communication

来源：评论

学校读者我要写书评

暂无评论

Determining polarity of speech signals based on gradient of spurious glottal waveforms

Determining polarity of speech signals based on gradient of ...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： Wen Ding N. Campbell ATR Interpreting Telecommunications Research Laboratories Soraku-gun Kyoto Japan

Speech polarity is crucial in many speech processing fields. We present a novel method to determine the polarity of speech signals from the gradient of spurious glottal waveforms. We use the iterative adaptive LPC inverse filtering to cancel the effect of vocal tract transfer function while maintaining most of the properties of source excitation. Then we take the first-derivative (gradient component) of spurious glottal waveforms to capture the sharp gradient near the glottal closure instant. By using the gradient components of the spurious glottal waveforms, we detect speech polarity, i.e., the polarity of glottal waveforms, by finding whether the glottal closure instants are located above or below the zero-line. Furthermore, a frame-based decision technique is applied to get robust results. Experimental results with a wide variety of speech utterances reveal a high performance and the computation complexity is much less than a previously proposed method.

关键词： Speech processing Speech synthesis Speech analysis Filtering Low pass filters linear predictive coding Adaptive filters Signal processing Signal analysis Signal synthesis

来源：评论

学校读者我要写书评

暂无评论

Variable model order LPC quantization

Variable model order LPC quantization

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： P. Ojala A. Lakaniemi Speech and Audio Systems Laboratory Nokia Research Center Tampere Finland

This paper presents a new method to apply variable bit-rate predictive quantization of the variable model order LPC parameters. In addition, the method is employed to interpolate the parameters within the analysis frame. The LPC model order selection algorithm is based on the characteristics of the input signal and on the performance of the LPC model. Hence, the variable bit-rate LPC quantization is source controlled. The number of quantized parameters needs to be identical in successive frames to be able to apply the predictive quantization and to interpolate parameters inside the frame. Therefore, the order of the LPC model of the previous frame needs to be expanded or reduced to be the same as the current frame LPC model. The advantage of variable model order LPC quantization is the lowered average bit-rate compared to a fixed rate while the speech quality remains the same.

关键词： linear predictive coding Quantization predictive models Speech synthesis Speech analysis Decoding Speech coding Speech codecs Cost function Testing

来源：评论

学校读者我要写书评

暂无评论

Wideband speech recovery from bandlimited speech in telephone communications

Wideband speech recovery from bandlimited speech in telephon...

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： H. Yasukawa Faculty of Information Science Aichi Prefectural University Aichi Japan

This paper describes methods that can enhance the quality of speech signals that are severely band limited during regular telephone speech transmission. We have already proposed a spectrum widening method that utilizes aliasing in sampling rate conversion and digital filtering for spectrum shaping. This paper discusses the method using linear prediction. Speech components of the outbands of the received signal are basically generated by LPC (linear predictive coding) synthesis by analysis. Furthermore, we discuss a new spectrum widening method using a multilayer backpropagation neural network. It is shown that the proposed method has a good performance of recovering the wideband speech.

关键词： Wideband Telephony Speech enhancement Speech synthesis Speech analysis linear predictive coding Multi-layer neural network Sampling methods Digital filters Filtering

来源：评论

学校读者我要写书评

暂无评论

A 1.9 kbps Zinc function excited, waveform interpolated speech codec

A 1.9 kbps Zinc function excited, waveform interpolated spee...

引用

IEEE Conference and Exhibition on Global Telecommunications (GLOBECOM)

作者： F.C.A. Brooks L. Hanzo Department of Electr Comp. Sc. University of Southampton UK

The Zinc function excited codecs of Hiotakakos and Xydeas (see Proceedings of the ICCS 94, p.865-9, 1994) and Sukar et al. (1989) were further developed by including a novel wavelet-based pitch-detector, by reducing its complexity and by eliminating the need for transmitting the Zinc function positions. The bit allocation of the table shown summarizes the most salient codec features.

关键词： Zinc Speech codecs Prototypes Interpolation Speech analysis Speech synthesis linear predictive coding Bit rate Speech coding Signal synthesis

来源：评论

学校读者我要写书评

暂无评论

A novel algorithm to estimate the line spectral frequencies from LPC coefficients

A novel algorithm to estimate the line spectral frequencies ...

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： M.R. Nakhai F.A. Marvasti Department of Electronic Engineering King''s College University of London London UK

A new method to estimate the line spectral frequencies (LSF's) with any desired accuracy is described. LSF's are estimated by locating the zeros of two polynomials derived from LPC inverse filter polynomial on the unit circle in z-plane. The computational load increases as the required accuracy increases. An important feature of algorithm is the ability of compromising the required accuracy and computational load. The method can also be applied to convert the higher order LPC coefficients to associated LSF's. The algorithm does not require a very high precision arithmetic and can be implemented using fixed-point digital signal processors.

关键词： Frequency estimation linear predictive coding Polynomials Filters Signal processing algorithms Speech analysis Speech coding Quantization Dynamic range Telecommunication computing

来源：评论

学校读者我要写书评

暂无评论

Fusion of auditory and visual information for noisy speech enhancement: a preliminary study of vowel transitions

Fusion of auditory and visual information for noisy speech e...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： L. Girin G. Feng J.L. Schwartz Institut de la Communication Parlée UPRESA 5009 INPG ENSERG Université Stendhal Grenoble France

This paper deals with a noisy speech enhancement technique based on the fusion of auditory and visual information. We first present the global structure of the system, and then we focus on the tool we used to melt both sources of information. The whole noise reduction system is implemented in the context of vowel transitions corrupted with white noise. A complete evaluation of the system in this context is presented, including distance measures, Gaussian classification scores, and a perceptive test. The results are very promising.

关键词： Speech enhancement Vectors linear predictive coding Filters Working environment noise Filtering Parameter estimation Speech synthesis linear regression Data mining

来源：评论

学校读者我要写书评

暂无评论

Design of Components for a 7-16 Precision Calibration Kit Using the High Frequency Structure Simulator

Design of Components for a 7-16 Precision Calibration Kit Us...

引用

ARFTG Conference

作者： Bela B. Szendrenyi Maury Microwave Corporation Ontario CA USA

A systematic method to design components of a 7-16 connector type precision short-open-load-thru (SOLT) calibration kit for the wireless industry using the High Frequency Structure Simulator (HFSS) at microwave frequencies is presented. To achieve a better than 60 dB accuracy in the nominal optimized value of each individual feature, only one of the 5° segments of the coaxial structure was used to model the equivalent 2D longitudinal cross-section. The validation process for this simplification, together with the test results on the components using a reference TRL/LRL calibration kit for the microwave measurements, is demonstrated. A better than 1.025 VSWR for the 7mm to 7-16 adapters up to 7.5 GHz is reported.

关键词： Calibration Connectors Testing linear predictive coding Design methodology Microwave frequencies Coaxial components Design optimization Production Microwave measurements

来源：评论

学校读者我要写书评

暂无评论

没有更多数据了...

全选清除本页清除全部题录导出标记到“检索档案”

共248页 << < 98 99 100 101 102 103 104 105 106 107 > >>

检索报告对象比较合并检索0

隐藏清空

合并搜索

回到顶部

执行限定条件

内容：

评分：

请选择保存的检索档案：

请选择收藏分类：

订阅名称：

通借通还

温馨提示：

图书名称：

借书校区：

取书校区：

手机号码：

邮箱地址：

一卡通帐号：

电话和邮箱必须正确填写，我们会与您联系确认。

联系人：

所在院系：

联系邮箱：

联系电话：

内蒙古自治区呼和浩特市赛罕区大学西街235号邮编: 010021

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：