speech signal

Through a process to enchance the noise speech signal, the SNR increased by 2& 7dB, with the result of its audibility improved.

这噪声信号通过增强处理后，使得信噪比提高了3～7dB，可听懂度得到改善。

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The results show that speech recognition rate can be enhanced by implementing pre-processing to noisy speech signal, and the speech recognition effect of sub-band is better than full-band in a way.

通过大量实验结果表明，对带噪语音信号进行有效的语音增强预处理能提高语音识别率，而且在某种程度上子带语音识别相对于全带语音识别效果更好一些。

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The pitch period and the spectral envelope of speech represent acoustical characteristics of the speech signal.

基音周期体现了发声过程肺部气流的周期性,而语音频谱包络是人的声道共振滤波特性在频域中的表现.

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The Design of Speech Signal Data Collection System and Interface Circuit With a PC/ XT Computer

语音信号采集系统及其与PC/XT机的接口电路的设计

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Speech preprocessing, including signal pre-emphasis, windowed and framed and end-point detection algorithm.

语音信号预处理：信号预加重、加窗与分帧、端点检测算法等研究与实现。

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Over the years, it has been the goal that compressing the transmission bandwidth of speech signal or reducing the transmission rate of telephone channel. In achieving this target, speech compression coding technology plays an important role.

多年来，人们一直追求的目标就是压缩语音信号的传输带宽或降低电话信道的传输码率，而在实现这一目标中，语音压缩编码技术发挥着重要作用。

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Compressed Speech Signal Sensing Based on Approximate KLT

基于近似KLT域的语音信号压缩感知

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Since the widely used Hidden Markov Model ( HMM) in speech recognition is first order Markov Model, it can not fully model the temporal dependence of speech signal.

由于在语音识别中被广泛应用的隐马尔可夫模型（HMM）是一重马尔可夫模型，它不能充分地描述语音信号的时间相依性。

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Speech signal is the most convenient and shortcut way of intercommunion.

语音对话是人们相互通讯和交流最方便快捷的手段.

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The speech signal was to be sampled for its 'levels' fifty times a second, which meant that the transmission was roughly equivalent to sending 300 teleprinter characters a second.

语音信号每秒要采样50次，这意味着每秒要发送300个字符。

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Parameter synthesis method is mature in theory and easy to be implemented, but the synthesized speech signal is not natural and has an obvious artificial feeling.

对参数合成法的研究起步比较早，所以参数合成法已具有相对成熟的理论基础，而且实现起来较简单，但实际合成的连续语音不自然，有明显的机器腔。

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This paper advances a new method for the noise reduction based on the difference function, analyzes on its principle, and illustrates the SNR ( signal-to-noise ratio) improving effect by SPAD ( speech processing system by using auto-correlation function).

提出了一种新的差分函数降噪方法，对其原理进行了分析，并对差分函数声音处理方式的信噪比改善效果进行了说明。

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With the digital and network of speech signal, VoIP is coming in our world.

随着语音信号数字化,网络化浪潮的到来, VoIP系统逐渐风靡.

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First the speech recognition related theories and methods are introduced from the signal preprocessing, windowing and framing, endpoint detecting, feature extraction, to the training and recognition steps.

从语音信号预处理、加窗分帧、端点检测、特征参数提取、识别和训练六个环节详细阐述了语音识别的原理、处理流程及相应算法。

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Elimination of the background noise in speech is one of the most challenging problems of speech signal processing.

剔除语音中的背景噪声是语音信号处理中具有挑战性的问题。

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In this paper, multi-parameters ( short-time energy, short-time zero-crossing rate and autocorrelation function values) combination has been adopted to judge the unvoiced/ voiced of speech signal. Then circular average magnitude difference function has been used to calculate pitch.

采用基于多参数组合方法（短时能量、短时过零率、自相关函数值）进行语音信号清浊音判别，利用循环平均幅度差函数进行基音周期计算。

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We have applied many complexity measures and theory of statistics to analyzing many speech signal.

本文运用多种复杂性算法和统计的理论分析了大量的语音信号.

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Voice conversion is a new branch of the speech signal processing, the technology can be used in text-to-speech system, film dubbing, secure communication and so on, it has very important research value.

语音转换是语音信号处理领域一个较新的分支，该技术可用于文语转换系统、电影配音和保密通信等多个领域，具有很重要的研究价值。

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Then the speech signal is pitch adjusted according to this pitch contour, where the original signal with time varying pitch period is converted to a signal of constant pitch period.

然后根据这一轨迹对语音信号进行基音调整，将原始的、具有时变基音周期的信号转化为一个具有恒定基音周期的信号。

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2-D adapting filter make original character of speech signal be kept.

而采用二维自适应滤波后，语音的包络形状得到较好地保持，使原始语音信号的特征完整。

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In the preprocessing stage, we add window and detect endpoint to the speech signal before framing, then we denoise the signal with the denoising model.

在预处理阶段的分帧加窗及端点检测之前，先用去噪模型去噪。

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Experiments show the efficiency of removing noisy signal in noisy speech signal by combining mathematical morphology and wavelet transform.

实验结果表明，数学形态学和小波变换两者相结合能够有效地滤除混噪语音信号中的噪声信号。

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Fast Realization of Long Signal Convolution Computation and Its Application to Speech Processing

长信号卷积的快速运算及其语音处理的应用

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Speaker recognition, also known as voiceprint recognition, is an important branch of speech signal processing and is one of the most popular research fields about speech.

说话人识别又称声纹识别，是语音信号处理中的重要组成部分，是当前的研究热点之一。

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With regard to communication security, the safekeeping of password and real-time processing in speech signal encryption based on triple Data Encryption Standard ( 3DES) were analyzed. In addition, the basic framework of speech encryption and the hardware system design in encryption telephone were presented.

针对通信安全性问题，分析了三重数据加密的密钥保管问题和语音加密的实时处理技术，提出了将算法移植到电话中加密语音信号的系统结构并进行了硬件设计。

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Therefore, the speech signal enhancement technique, automatic control engineering and room acoustics are used to solve the very problem the noise, interference problem in the voice control technique for household appliances In this paper, the adaptive noise cancelling ( ANC) system is designed.

指出，电视机等电器发出的语声，是最为严重的干扰噪声。为此，本文引入ANC方法这一有效的语音增强技术，提出了一种用于家用电器声控制技术的干扰噪声消除方案。

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Yinmarkerf development kit for the analysis and recognition of speech signal.

隐马尔科夫工具开发包,用于语音信号的分析和识别.

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The speech signal is essentially continuous, so the speaker recognition rate can be improved if we obtain the temporal correlation in the frames.

语音信号本质上具有连续性，所以获取语音帧特征之间的时变特性，将对识别的性能带来提高。

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Based on traditional attenuation factor method, the short-term energy and the short-term zero crossing rate of the speech signal is used to dynamically revise the mixing factor in the proposed method.

该算法在传统的混音衰减因子法的基础上，采用短时能量及短时过零率对混音的衰减因子进行动态修正。

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Proposal and implementation of speech signal feature analysis and related researches.

建议和实施的语音信号特征分析及相关研究.

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