Description: SFS提供了计算语音本质的一系列工具。包括支持各种数据格式，信号处理和图形，频谱、共振峰分析，基频提取等等。在线文档说明请访问http://www.phon.ucl.ac.uk/resource/sfs/help/-SFS provides a voice calculated nature of a series of tools. Including support for various data formats, signal processing and graphics, spectrum, formant analysis, frequency, etc. extraction. Online document shows, please visit http : / / www.phon.ucl.ac.uk/resource/sfs/help/ Platform: |
Size: 1294860 |
Author:lixiao |
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Description: 语音情感是别中经常用到的基频F0和前三个共振峰F1F2F3的提取程序，并把引用的辅助函数nanmax.m，nanmean.m，nanmin.m一并附上。直接调用demo01.m即可。-emotional voice is not often used in the fundamental frequency F0 and the former three formant F1F2F3 the process of extraction, and the auxiliary function invoked nanmax.m, nanmean.m. nanmin.m accompanied. Demo01.m can call directly. Platform: |
Size: 523880 |
Author:lijin |
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Description: Speech analysis and parameter extraction Short-term analysis, frames and windows Time-domain analysis: energy, zero-crossings, statistic parameters, autocorrelation
Frequency-domain analysis: spectra and spectrograms Cepstral analysis Linear prediction analysis Pitch and formant estimation
to run the application please tape menu
Platform: |
Size: 2220656 |
Author:栗子 |
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Description: SFS提供了计算语音本质的一系列工具。包括支持各种数据格式，信号处理和图形，频谱、共振峰分析，基频提取等等。在线文档说明请访问http://www.phon.ucl.ac.uk/resource/sfs/help/-SFS provides a voice calculated nature of a series of tools. Including support for various data formats, signal processing and graphics, spectrum, formant analysis, frequency, etc. extraction. Online document shows, please visit http :// www.phon.ucl.ac.uk/resource/sfs/help/ Platform: |
Size: 1294336 |
Author:lixiao |
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Description: 语音情感是别中经常用到的基频F0和前三个共振峰F1F2F3的提取程序，并把引用的辅助函数nanmax.m，nanmean.m，nanmin.m一并附上。直接调用demo01.m即可。-emotional voice is not often used in the fundamental frequency F0 and the former three formant F1F2F3 the process of extraction, and the auxiliary function invoked nanmax.m, nanmean.m. nanmin.m accompanied. Demo01.m can call directly. Platform: |
Size: 523264 |
Author:lijin |
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Description: Speech analysis and parameter extraction Short-term analysis, frames and windows Time-domain analysis: energy, zero-crossings, statistic parameters, autocorrelation
Frequency-domain analysis: spectra and spectrograms Cepstral analysis Linear prediction analysis Pitch and formant estimation
to run the application please tape menu
Platform: |
Size: 2220032 |
Author:栗子 |
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Description: 本工具箱利用lpc方法可以准确的提取出共振峰频率，并且解决了共振峰检测时滤波器阶数对所有说话人都一样的问题。本工具箱由美国卡内基梅隆大学一位教授编写，因此所有解释均为英文原版。-The toolbox to use LPC method can extract accurate formant frequency, and the resolution of formant filter order testing of all the speaker are the same problems. The toolbox by the United States a professor at Carnegie Mellon University to prepare, so all the explanations are the original English edition. Platform: |
Size: 44032 |
Author:yanjingyu |
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Description: this program uses an impulse train, takes formant frequencies and filters them. output signal is synthesized sound.
commented part ,controls time=d, sampling frequency=fs
normal frequency=f
and bits=n, then synhesizes the output sound Platform: |
Size: 1024 |
Author:G眉ven |
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Description: the code i submitted runs through matlab, this code calculates formant frequency, fundamental frequency & picth of the feed voice sample,
after running the code it should be feed with 4 voice samples of exactly 2 sec each Platform: |
Size: 2048 |
Author:ananth |
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Description: 铺的峰值基本上对应于共振峰频率，对平滑过的对数谱中的峰值进行定位即可估计共振峰-Shop peak is basically corresponds to the frequency of the resonance peak can be estimated, smoothed to locate the peak in the number of spectral formant Platform: |
Size: 1024 |
Author:fengchailiu |
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Description: 基于MATLAB用于提取语音信号的特征参数如倒谱，语谱等，可以方便的检测出语音信号的基频，共振峰。-Cepstrum based on MATLAB is used to extract the characteristics of the speech signal parameters such as language spectrum, can easily detect the speech signal fundamental frequency, formant. Platform: |
Size: 1024 |
Author:王毅 |
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Description: 语音信号进行分析、处理和设计。用MATLAB做变声器的原理是通过改变输入声音频率，进而改变声音的音色、音调，使输出声音在感官上与原声音不同。变声器是借助对声音音色和音调的双重复合改变，实现输出声音的改变。通过自己发声，共振峰频率的改变是基于重采样实现的-Speech signal analysis, processing and design. MATLAB do voice-changer works by changing the input frequency of the sound, and then change the sound of the tone, pitch, so that the output audio with the original sound in different senses. Voice Changer is the use of sound timbre and tone of the dual composite changes, realize the output sound changes. Through their own sound, formant frequency change is implemented based on resampling Platform: |
Size: 10240 |
Author:刘安兰 |
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Description: 语音信号进行分析、处理和设计。用MATLAB做变声器的原理是通过改变输入声音频率，进而改变声音的音色、音调，使输出声音在感官上与原声音不同。变声器是借助对声音音色和音调的双重复合改变，实现输出声音的改变。通过自己发声，共振峰频率的改变是基于重采样实现的-Speech signal analysis, processing and design. MATLAB do voice-changer works by changing the input frequency of the sound, and then change the sound of the tone, pitch, so that the output audio with the original sound in different senses. Voice Changer is the use of sound timbre and tone of the dual composite changes, realize the output sound changes. Through their own sound, formant frequency change is implemented based on resampling Platform: |
Size: 7168 |
Author:刘安兰 |
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Description: 语音处理是信号与信息处理的重要内容之一，通过本课程设计，使学生理解数字信号处
理的有关理论和方法在语音处理中的具体应用。课程设计的目的归纳如下：
1、掌握语音信号的特点；
2、掌握语音处理的基本理论和方法；
3、掌握基于Matlab编程实现语音的获取、显示、频谱分析、短时能量、短时自相关以及倒
谱复倒谱的分析方法；
4、掌握语音基音频率及共振峰频率的检测方法。-Signal and speech processing is an important part of information processing, through the curriculum design, so that students understand the digital signal
The theory and methods of management in voice processing in the specific application. Curriculum is designed to be summarized as follows:
A master speech signal characteristics
2, master the basic theory and speech processing methods
3, master Matlab programming based voice access, display, spectrum analysis, short-time energy, short-term autocorrelation and inverted
Spectral analysis of the complex cepstrum method
4, master voice pitch frequency and formant frequency detection method.
Platform: |
Size: 1226752 |
Author:立枣酒 |
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Description: Abstract—Several algorithms have been developed for tracking
formant frequency trajectories of speech signals, however most of
these algorithms are either not robust in real-life noise environments
or are not suitable for real-time implementation. The algorithm
presented in this paper obtains formant frequency estimates
voiced segments of continuous speech by using a time-varying
adaptive filterbank to track individual formant frequencies. The
formant tracker incorporates an adaptive voicing detector and a
gender detector for formant extraction continuous speech,
for both male and female speakers. The algorithm has a low signal
delay and provides smooth and accurate estimates for the first four
formant frequencies at moderate and high signal-to-noise ratios.
Thorough testing of the algorithm has shown that it is robust over
a wide range of signal-to-noise ratios for various types of background
noises.-Abstract—Several algorithms have been developed for tracking
formant frequency trajectories of speech signals, however most of
these algorithms are either not robust in real-life noise environments
or are not suitable for real-time implementation. The algorithm
presented in this paper obtains formant frequency estimates
voiced segments of continuous speech by using a time-varying
adaptive filterbank to track individual formant frequencies. The
formant tracker incorporates an adaptive voicing detector and a
gender detector for formant extraction continuous speech,
for both male and female speakers. The algorithm has a low signal
delay and provides smooth and accurate estimates for the first four
formant frequencies at moderate and high signal-to-noise ratios.
Thorough testing of the algorithm has shown that it is robust over
a wide range of signal-to-noise ratios for various types of background
noises. Platform: |
Size: 67584 |
Author:assouma |
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Description: 基于Hilbert-Huang变换的语音信号共振峰频率估计，一种对语音信号的分析方法-Based on the Hilbert Huang transform of speech signal formant frequency estimation, a method on the analysis of the speech signal Platform: |
Size: 430080 |
Author:娜娜 |
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Description: The espeakedit program is used to prepare phoneme data for the eSpeak speech synthesizer.
It has two main functions:
Prepare keyframe files for individual vowels and voiced consonants. These each contain a sequence of keyframes which define how formant peaks (peaks in the frequency spectrum) vary during the sound.
Process the master phonemes file which, by including the phoneme files for the various languages, defines all their phonemes and references the keyframe files and the sound sample files which they use. espeakedit processes these and compiles them into the phondata, phonindex, and phontab files in the espeak-data directory which are used by the eSpeak speech synthesizer.-espeak edit
for TTS Platform: |
Size: 4963328 |
Author:YKLin |
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Search - formant frequency