Description: 这是一个语音聊天的源代码,完成了主要的功能,只是采集的语音信号没有编码就直接发送出去了,有兴趣的朋友可以-This a voice chat source code, to complete the main function, but the acquisition of the voice signal coding not directly sent away, interested friends can s Platform: |
Size: 659106 |
Author:于哲 |
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Description: 语音信号处理方面的源代码,简单的语音合成软件(text to speech)-voice signal processing of the source code, a simple speech synthesis software (text to speech) Platform: |
Size: 9388 |
Author:玉贤哲 |
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Description: 语音压缩编码源程序LPC10。用于语音信号的压缩、编码、增强等-voice coding source LPC10. For voice signal compression, encoding and so on Platform: |
Size: 562205 |
Author:蒋荣 |
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Description: 这是一个语音聊天的源代码,完成了主要的功能,只是采集的语音信号没有编码就直接发送出去了,有兴趣的朋友可以-This a voice chat source code, to complete the main function, but the acquisition of the voice signal coding not directly sent away, interested friends can s Platform: |
Size: 658432 |
Author:于哲 |
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Description: 在语音信号处理中使用matlab对语音信号进行lpc谱估计的源代码。-in voice signal processing using Matlab, the audio signal lpc spectrum estimation of the source code. Platform: |
Size: 1024 |
Author: |
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Description: 用MATLAB来实现语音信号处理HMM的一些函数的源程序。-MATLAB to voice signal processing functions HMM some of the source. Platform: |
Size: 254976 |
Author:W |
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Description: 语音合成程序!psalo频域基音同步叠加方法。它首先对原始语音信号进行短时频域变换,得到短时谱和短时谱包络。短时谱除以短时谱包络得到声源短时谱,对声源短时谱的实部和虚部分别进行线性插值,就可以达到改变语音信号基频的目的,然后再进行频域反变换,可得到变换后的短时语音信号。短时谱包络部分也可以独立改变,以达到改变音色的目的。-speech synthesis procedures! Psalo frequency domain pitch synchronous superposition method. It is first of the original speech signal for short-time frequency domain transform, to be short-term and short-term spectral envelope spectrum. Short-term spectrum divided by the short-time spectral envelope of the sound source to be short-term spectrum of the sound spectrum of short-term real and imaginary parts of linear interpolation, they could change the voice-frequency signals to the end, then we will proceed to frequency-domain transform, Transform available after the short speech signal. Short-term spectral envelope can be independent of changes to achieve the objective of changing colors. Platform: |
Size: 2048 |
Author:hzh |
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Description: 读取语音信号(用matlab的wavread指令),把语音信号分帧、加窗,进行清浊分割,提取基
频,这一部分较简单,自己编程序做。参考文献自己到图书馆期刊网上查找。
提取语音信号的lpc参数,可调用lpcfit.m 程序(我提供,见附件),将源、目标语音的浊音
段的lpc系数进行DTW规整,调用pathita2.m 程序(我提供,见附件)。将规整得到的lpc系数
转换成lsp参数,调用lpcar2ls.m 程序(我提供,见附件), 再进行转换映射,调用matlab
的指令newrbe。-read the speech signal (using Matlab wavread Directive), Voice Signal frames, Windowed, Qingzhuo segmentation, extracting fundamental frequency, this part is relatively simple, they programmed to do. References their online journals to the library search. Voice Signal Extraction lpc parameters can be called lpcfit.m (I provide, see Annex), the source, Voiced objective voice of the lpc coefficient DTW structured, Call pathita2.m (I provide, see Annex). Structured to the lpc lsp conversion coefficient parameters, calling lpcar2ls.m (I offer see Annex), the shift mapping, called the directive newrbe Matlab. Platform: |
Size: 4096 |
Author:韦敏 |
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Description: 语音信号处理软件praat教程:这是一部入门教程,引导您学习Praat——可以分析、合成、变换语音并为论文著述创建优质图表的计算机程序。-Voice signal processing software praat Tutorial: Getting Started This is a tutorial to guide you to learn Praat- can be analyzed, synthesis, voice transformation and graph paper to create quality writing computer programs. Platform: |
Size: 171008 |
Author:朱建伟 |
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Description: 语音信号端点检测方面的源程序,比较实用.-Voice signal endpoint detection of the source area, more practical. Platform: |
Size: 2313216 |
Author:zhuying |
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Description: 盲信号分离(BSS)是指在对彼此独立的源信号混合过程及各源信号本身均未知的情况下,从混合信号中分离出这些源信号的方法。BSS可以用来从多个麦克风混合语音信号中提炼出单个语音信号。本文简要阐述LMS、RLS算法,并通过仿真实验来分析比较这两类方法的性能,并利用此方法对一实际的语音信号进行分离。-Blind Signal Separation (BSS) is defined as independent of each other mixed-signal process and the source of the signal itself are unknown circumstances, from the mixed-signal to isolate the source of these signals. BSS can be used from multiple microphones mixed voice signal to extract a single speech signal. This paper briefly described LMS, RLS algorithm, and through simulation experiments to analyze the comparative performance of these two types of methods and take advantage of this method on a real speech signal separation. Platform: |
Size: 1164288 |
Author:贡晓飞 |
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Description: 用于语音信号分析的matlab源代码,包括谐波分析,声强分析,音源分析等
This software consists of a series of Matlab routines useful for
various aspects of prosodic analysis of speech, in particular F0
and harmonic analysis, loudness analysis, and voice-source analysis. -For voice signal analysis matlab source code, including harmonic analysis, sound intensity analysis, audio analysis, etc. This software consists of a series of Matlab routines useful forvarious aspects of prosodic analysis of speech, in particular F0and harmonic analysis, loudness analysis, and voice-source analysis. Platform: |
Size: 72704 |
Author:凝空子 |
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Description: this a source code which can use for encryption of voice signal by using the kasami sequence.Kasami sequence is a pseudo random sequence. -this is a source code which can use for encryption of voice signal by using the kasami sequence.Kasami sequence is a pseudo random sequence. Platform: |
Size: 2048 |
Author:shanaka |
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Description: 录制一段个人自己的语音信号,并对录制的信号进行采样;画出采样后语音信号的时域波形和频谱图;给定滤波器的性能指标,采用窗函数法和双线性变换设计滤波器,并画出滤波器的频率响应;然后用自己设计的滤波器对采集的信号进行滤波,画出滤波后信号的时域波形和频谱,并对滤波前后的信号进行对比,分析信号的变化;回放语音信号-The individual' s own record a voice signal, and the recorded signal is sampled draw sampled speech signal time-domain waveform and frequency spectrum filter performance given by the window function method and bilinear transformation design a filter and draw the filter frequency response then use their own filters designed to filter the collected signals, to draw the filtered time domain waveform and frequency spectrum, and comparing the signal before and after filtering, analysis of signal changes playback of the speech signal Platform: |
Size: 12288 |
Author:姚湘陵 |
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Description: 基于MATLAB语音信号采集与分析加上高频噪声并去噪声源程序。-Based on MATLAB voice signal acquisition and analysis with high-frequency noise to the noise source. Platform: |
Size: 911360 |
Author:majp |
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Description: 本书系统地阐述了语音信号处理的原理、方法、技术和应用,同时给出了部分内容对应的MATLAB仿真源程序。全书共12章,第1章至第7章是基本理论部分,包括语音信号的数字模型、语音信号的短时时域分析和频域分析、语音信号的同态处理、语音信号线性预测分析和矢量量化;第8章至第12章是应用部分,包括语音编码、语音合成、语音识别、语音增强和语音处理的实时实现。本书内容全面,重点突出,原理阐述深入浅出,注重理论与实际应用的结合,可读性强。-This book describes the speech signal processing principles, methods, techniques and applications, and gives the corresponding part of the contents of the MATLAB simulation source. The book is 12 chapters, Chapter 1 to Chapter 7 is the basic theoretical part, including voice signal digital model, speech signal analysis in time domain and frequency domain analysis, speech signal homomorphic processing, speech signal analysis and linear prediction vector quantified Chapter 8 to Chapter 12 is the application of parts, including speech coding, speech synthesis, speech recognition, speech enhancement and voice processing, real-time implementation. The book is comprehensive, focused and Rationale layman, focusing on the combination of theory and practical application, readable. Platform: |
Size: 15129600 |
Author:Qin |
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Description: 对语音信号进行采样率的转换,如整数倍的内插和抽取,还可以进行非整数倍的采样率转换。-The sampling rate of the voice signal conversion, such as interpolation and decimation integer multiple of, can also be a non-integer multiple of the sampling rate conversion. Platform: |
Size: 5120 |
Author:张辉 |
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