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new speech
相关语句
  新的语音
     A New Speech Recognition System with DSP
     用DSP技术开发的一种新的语音识别系统
短句来源
     DAST-A New Speech Processing Technique
     DAST──一种新的语音处理技术
短句来源
     Using Mallat algorithm on the basis of wavelet transform and multiresolution analysis(MRA) in connection with traditional FFT,a new speech signal fundamental frequency detection method is presented.
     将基于小波变换与多分辨分析 (MRA)下的Mallat算法与传统的快速傅立叶变换(FFT)相结合 ,提出了一种新的语音信号的基音频率检测方法。
短句来源
     A New Speech Synthesis Algorithm Based on MBE
     基于MBE算法的一种新的语音合成方法
短句来源
     A new speech recognition method by the integration of the VQ, HMM is proposed.
     提出了一种新的语音识别模型VQ-HMM。
短句来源
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  新话语
     Finding the New Speech to the Modern Research of Local Governance
     寻找当代地方治理研究的新话语
短句来源
  新型语音
     A New Speech Processing System
     一种新型语音处理系统
短句来源
     The Realization of Algorithm of New Speech Communication System Based on G.729 Algorithm
     基于G.729算法的新型语音通信系统的实现
短句来源
     In this paper, we give a new speech coding algorithm based on ADPCM.
     为了实现本论文的目的,本论文设计了一种基于ADPCM 编码算法的新型语音编解码算法。
短句来源
     This paper presents a new speech codec which will be used in3G mobile communication systems and packet networks. A bandwidth of50to7000Hz improves the intelligibility and naturalness of speech so it is suitable for multimedia services for third generation mobile communication systems ,wideband telephony over packet networks,au-dio and video teleconferencing.
     文章介绍了一种应用于第三代移动通讯的新型语音编解码器,该编解码器的语音带宽拓展为50Hz到7000Hz,编码后语音的自然度很高,用在3G移动通讯系统的多媒体服务、宽带包交换网络、音频和视频会议等等。
短句来源
     This paper presents a new speech codec-Variable Rate Multimode Wideband codec(VMR-WB) which will be used in 3G mobile communication system CDMA2000 and packet networks.
     本文介绍了一种将应用于第三代移动通信的新型语音声码器-变速率多模式宽带(VMR-WB)声码器。
短句来源
  “new speech”译为未确定词的双语例句
     A New Speech Coder Standard──G.729
     新一代语音编码标准──G.729
短句来源
     A New Speech Recognition System Using SOM and MLP Neural Networks
     SOM结合MLP的神经网络语音识别系统
短句来源
     A new speech coding scheme at 2kbit/s was proposed based on traditional waveform interpolation (WI) speech coding model.
     基于传统的波形内插语音编码模型,提出了一种新的2kbit/s语音编码方案。
短句来源
     A New Speech Segmentation Algorithm Based on Wavelet Transform and Parametric Filter
     基于小波变换和参数滤波的音素分段算法
短句来源
     By the experiment, we could conclude that the new speech recognition and understanding model with close ring structure are effective and reasonable.
     本文在分析了人对自然语音识别理解的特点之后,提出了这个新的闭环语音识别理解模型,然后把该模型与传统模型进行对比分析,最后就该模型进行具体的汉语语音识别理解实验,并对实验结果进行了总结。
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  new speech
Four patients (33%) needed a second surgical procedure, and in one patient (8%) inserting a new speech prosthesis obliterated the residual fistula.
      
We would like to find a sequence of triphone videos from our database that matches this new speech utterance.
      
Unvoiced speech sounds are additively combined with the modified harmonic and aspiration noise components to yield the new speech signal.
      
The top window displays the new speech file, while the bottom window displays the old speech file.
      
The performance of the new speech recognizer is compared with that of a system constructed using conventional generalized triphone models.
      
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At the present time, the newest speech processing devices are made up of bit slicespeedy microprocessors and hardware multipliers. All operations are governed by software. This paper describes a new scheme of speech processing device employ ag multiple low speed microprocessors like Intel 8080 A, 8085 or Motorola 6800, and also gives specific method of solution, based on experiments with Intel 8080A, 8085 hardware and software, to the key problems of implementation with multiple microprocessors,...

At the present time, the newest speech processing devices are made up of bit slicespeedy microprocessors and hardware multipliers. All operations are governed by software. This paper describes a new scheme of speech processing device employ ag multiple low speed microprocessors like Intel 8080 A, 8085 or Motorola 6800, and also gives specific method of solution, based on experiments with Intel 8080A, 8085 hardware and software, to the key problems of implementation with multiple microprocessors, such as enhancing the computing speed, communication between microprocessors and the related interfacing technique, system management, hardware and software differentiation, etc. This constitution is of value to extend speedily the use of nicroprocescors in the field of digital communication in our country. With the development of MOS microprocessors, it will be capable of competition in the long run.

目前最新的语言信号处理机都用位片式快速微处理器加硬件乘法器构成,全部操作用软件实现。本文提出了由多个速度较低的微处理器如Intel公司的8080A、8085或Motorola公司的6800构成语言处理机的方案。并且在对硬件和软件进行实验的基础上,对多机系统的一些关键问题如提高运算速度的措施、微处理器之间的通信与有关接口技术、系统的管理、硬件软件的划分等,提出了具体的方法。这种体制对于目前在国内的数字通信领域中迅速推广使用微处理器是有价值的。从长远看,随着MOS型微处理器的发展,也将会有竞争能力。

A new speaker-dependent speech recognition strategy based on Chinese syllable is discussed, and a feature abstraction approach is introduced, in which a new speech parameter based on auditory characteristic can be abstracted. This parameter of spectrum variance can reflect the character of Chinese speech concerned with speech perception. The syllable is selected to be used as the basic unit for recognition. The word vocabulary consists of 400 basic Chinese syllable. The result of experiment...

A new speaker-dependent speech recognition strategy based on Chinese syllable is discussed, and a feature abstraction approach is introduced, in which a new speech parameter based on auditory characteristic can be abstracted. This parameter of spectrum variance can reflect the character of Chinese speech concerned with speech perception. The syllable is selected to be used as the basic unit for recognition. The word vocabulary consists of 400 basic Chinese syllable. The result of experiment using this recognition strategy is presented.

本文探讨了一种特定人的汉语全音节语音识别方案,介绍了一种基于人耳听觉特性的语音参数的提取方法,对以1/3倍频程分布的16个通道滤波器组的对数能量输出用非线性时域归正方法归正到定长,然后求出相邻通道间频谱的变化量,即得到一组新的特征参数——频变参数.这组参数能够较好地反映语音中与感知有关的特性,如高音、音强、音调等.音节被选用来作为识别的基本单位,以400个汉语无调音节作为字表.最后给出了识别结果.

This paper suggests a new speech modeling technique-Multisequence Excited LPC (MSLPC). In this model, the excitation is represented by a set of limited sequences with long-term intervals. The long-term feature involved in excitation enables effective utilization of long-term redundancy in voiced speech and also leads to the development of a fast search algorithm for optimizing excitation parameters.After a brief introduction of the concept of MSLPC modeling, the block mode maximum crosscorrela-tion...

This paper suggests a new speech modeling technique-Multisequence Excited LPC (MSLPC). In this model, the excitation is represented by a set of limited sequences with long-term intervals. The long-term feature involved in excitation enables effective utilization of long-term redundancy in voiced speech and also leads to the development of a fast search algorithm for optimizing excitation parameters.After a brief introduction of the concept of MSLPC modeling, the block mode maximum crosscorrela-tion fast search algorithm of excitation parameters is discussed. Two kinds of product code VQ structures are designed and compared. Extensive experiments show that the MSLPC can be used to implement low bit rate speech coding within the range from 2.4kb/s to 4.8kb/s.

本文研究了一种新型的语音多序列激励线性预测(MSLPC)模型。它将激励用有限个长时规则序列表示。MSLPC激励模型中的长时特征使它能较好地利用语音浊音中的长时多余度,并能推出快速搜索算法。 本文在介绍了MSLPC模型的基本概念后,重点讨论了MSLPC激励参数的块模式最大互关快速搜索算法,以及激励序列幅度矢量的两种乘积码矢量量化(VQ)的设计。实验结果表明,MSLPC模型能用于实现在2.4 kb/s至4.8 kb/s速率段的低速率语音编码。

 
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