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[Speech/Voice recognition/combineenframe

Description: 对连续语音信号进行分帧的matlab程序-of continuous speech signal-frame Matlab procedures
Platform: | Size: 1024 | Author: wh | Hits:

[Speech/Voice recognition/combinewavetoframe1

Description: wav文件的与处理,包括预加重,分帧,合并,测试通过-wav files with the treatment, including pre-emphasis, sub-frame, combined, tested through
Platform: | Size: 91136 | Author: 尤凤凯 | Hits:

[matlabPreAndDeemphase

Description: 前面搞错了,把五个文件一起打包上传了,现在一个一个传 预加重声波在噪声信道内传输 模拟程序 程序头上有详细的注释和使用说明-Made a mistake in front of the pack with five file upload, and now a mass of a pre-emphasis acoustic noise within the transmission channel simulation program procedures detailed head notes and use
Platform: | Size: 9216 | Author: Hui Liu | Hits:

[Software Engineeringpre-emphasissub-framewindowed

Description: matlab编写的关于语音信号预加重,分帧加窗的程序。-matlab, prepared by the speech signal pre-emphasis, sub-frame windowed program.
Platform: | Size: 1024 | Author: 张艺 | Hits:

[Graph Recognizeshuohuarenshibie-MATLAB-duandianjiance-yujiazhong-

Description: 说话人识别代码,端点检测,预加重,MFCC..采用的模型是神经网络~-Speaker identification code, endpoint detection, pre-emphasis, MFCC .. neural network model is adopted ~
Platform: | Size: 5120 | Author: | Hits:

[File Formatvoice

Description: 语音的预加重和去加重,进行语音的前期处理,减少噪声。-Voice of the pre-emphasis and de-emphasis, voice of the pre-processing to reduce noise.
Platform: | Size: 5120 | Author: mengxiangai | Hits:

[matlabmatlab

Description: 语音识别预处理matlab实现,包括端点检测,预加重,加窗-Speech recognition pre matlab implementation, including endpoint detection, pre-emphasis, windowing
Platform: | Size: 1024 | Author: | Hits:

[matlabmeierdaopufadematlabshixian

Description: 对录音信号集 中的某一语音,利用BATLAB设计一美尔例谱算法,并实现。 取信号集 中的一个语音信号:“xxxxxx”,将它作为输入的语音信号来为设计一个美尔倒谱算法,在该算法中,主要设计了以下环节: 1.读入一个语音信号;2.对这个信号归一化;3.对归一化的信号进行加窗处理(这里的矩形窗长度必须为257,重帧长64);4.进行预加重处理,即通过一个高通滤波器: ;5进行512点的FFT;6.分别取模平方得到功率谱;7.在设计的mel滤波器组中,我采用了25个带通滤波器;8.将得到的功率谱信号通过mel滤波器组,得到相应的25个功率值;9.对这些功率值取自然对数;10.对这些值取离散作弦变换;11.将得到的值去掉直流分量,取其它值作为MFCC参数。 -Concentrated on recording a voice signal, using BATLAB designed Mayer cases spectral algorithm, and implementation. Get a voice signal signal focus: "xxxxxx", will it as input for the design of a voice signal to Mel Cepstrum algorithm in the algorithm, the main design of the following links: 1. Read a speech signal 2. On the signal normalized 3. Of the normalized windowed signal processing (here, rectangular window length must be 257, re-frame size 64) 4. For Pre-emphasis processing through a high-pass filter: 5 for 512 points of FFT 6. modulo square were obtained spectrum 7. in the design of mel filter banks, I use a 25 band-pass filter 8 . will be the power spectrum signal by mel filter bank, corresponding to the 25 power values 9. on the power to take the natural logarithm value 10. to take these values as strings of discrete transform 11. will be of value to out DC component, to take other values as MFCC parameters.
Platform: | Size: 13312 | Author: 赵欣 | Hits:

[matlabwindow-framing-matlab-program

Description: 语音信号处理预加重加窗分帧matlab程序-Speech signal processing pre-emphasis window framing matlab program
Platform: | Size: 1024 | Author: 宋涛 | Hits:

[assembly languageyuchuli

Description: 运用Matlab对语音文件进行预处理,预加重,滤波,去噪等处理-Using Matlab to a voice file preprocessing, pre-emphasis, filtering, denoising processing, etc
Platform: | Size: 11264 | Author: 张露 | Hits:

[Other9MATLABCHULIXIN

Description: 第9章共振峰的估算方法259 9.1预加重和端点检测259 9.1.1预加重259 9.1.2端点检测260 9.2倒谱法对共振峰的估算260 9.2.1倒谱法共振峰估算的原理260 9.2.2倒谱法共振峰估算的MATLAB程序261 9.3LPC法对共振峰的估算262 9.3.1LPC法共振峰估算的原理262 9.3.2LPC内插法共振峰的估算263 9.3.3LPC求根法共振峰的估算266 9.4连续语音LPC法共振峰的检测268 9.4.1简单LPC共振峰检测268 9.4.2改进的LPC共振峰检测270 9.5基于HilbertHuang变换(HHT)的共振峰检测274 9.5.1希尔伯特变换275 9.5.2语音信号的另一种模型——AMFM模型278 9.5.3对AMFM模型的分析279 9.5.4语音信号共振峰特征参数提取的HHT方法279 9.5.5基于HilbertHuang变换的共振峰检测步骤和MATLAB程序280-Estimation Chapter 9 259 9.1 formant pre-emphasis and pre-emphasis endpoint detection 259 259 9.1.1 9.1.2 9.2 endpoint detection principle cepstrum estimate 260 to 260 9.2.1 formant formant estimation cepstrum 260 9.2.2 cepstrum estimate estimate estimate formant MATLAB program 261 9.3LPC method formant 262 9.3.1LPC law principle of formant estimation interpolation within 262 9.3.2LPC formants 263 9.3.3LPC Root Law resonance estimate peak 266 9.4 Continuous Speech LPC formant detection method is simple LPC formant 268 9.4.1 268 9.4.2 Improved detection LPC 270 9.5 formant detection based HilbertHuang transform (HHT) 274 9.5.1 formant detection Hill Another model 275 9.5.2 Hilbert transform voice signals- AMFM model 278 9.5.3 Analysis of the AMFM model HHT Method 279 9.5.4 formant speech signal feature extraction based HilbertHuang transform 279 9.5.5 Resonance peak detection step and MATLAB program 280
Platform: | Size: 10240 | Author: 孟稳 | Hits:

[Communication-Mobilekikok

Description: 语音信号的分帧加窗的matlab实现及语音信号处理预加重、加窗分帧matlab程序-Sub-frame window of the speech signal and speech signal processing matlab to achieve pre-emphasis, adding windows framing matlab program
Platform: | Size: 16384 | Author: 卓鹏 | Hits:

[LabViewvoicebox

Description: 基于matlab采集语音信号;实现语音预加重、过零检测、分帧等预处理;实现可视化界面设计,完成语音预处理过程显示;-Matlab based speech signal acquisition, voice pre emphasis, zero detection, sub frame, etc. preprocessing visual interface design, complete voice preprocessing process display
Platform: | Size: 598016 | Author: 黄程 | Hits:

[Audio programgetwav2.m

Description: 用matlab的audiorecorder进行音频提取,声音预加重处理并画波形图-The audiorecorder with matlab for audio extraction, audio pre-emphasis and draw waveforms
Platform: | Size: 1024 | Author: youxin | Hits:

[Windows Developchapter3_1

Description: 对语音信号的预加重,实现了高频的提升的matlab程序-Pre-emphasis of speech signal, realized the high frequency of the matlab program of ascension
Platform: | Size: 1024 | Author: zandanfei | Hits:

[Embeded-SCM DevelopSTM32-Speech-Recognition-Master

Description: 于市售 STM32 开发板上实现特定人语音识别处理项目。识别流程是:预滤波、ADC、分帧、端点检测、预加重、加窗、特征提取、特征匹配。端点检测(VAD)采用短时幅度和短时过零率相结合。检测出有效语音后,根据人耳听觉感知特性,计算每帧语音的 Mel 频率倒谱系数(MFCC)。然后采用动态时间弯折(DTW)算法与特征模板相匹配,最终输出识别结果。先用Matlab对上述算法进行仿真,经数次试验求得算法内所需各系数的最优值。而后将算法移植到 STM32 开发板上,移植过程中根据 STM32 上存储空间相对较小、计算能力也相对较弱的实际情况,对算法进行优化。最终完成于 STM32 微处理器上的特定人语音识别系统。-Implement speech recognition processing project in commercially available STM32 development board. Identification is the process: pre-filter, ADC, framing, endpoint detection, pre-emphasis, windowing, feature extraction, feature matching. Endpoint detection (VAD) short-time amplitude and short-term zero rate combined. After detecting an effective voice, according to the characteristics of human auditory perception, calculated for each frame of speech Mel Frequency Cepstral Coefficients (MFCC). Then dynamic time warping (DTW) algorithm and feature template matches the final output recognition result. First with Matlab simulation algorithm described above, after several trials to get the optimal value of each coefficient within the desired algorithm. The algorithm will migrate to STM32 development board, the porting process according to the STM32 relatively small storage space, computing power is relatively weak situation of the optimization algorithm. Finally completed on the STM32 micr
Platform: | Size: 325632 | Author: Chenkly | Hits:

[Speech/Voice recognition/combineJLDATA

Description: 摘 要:本论文主要研究了语音识别的基本原理,对语音识别系统的构成进行分析处理,其中包括预处理、特征参数提取、建立模块库、识别匹配几大部分。预处理又包括语音采样、预加重、加窗(汉明窗)、端点检测;特征提取的参数是梅尔频率倒谱系数MFCC。 该语音系统采用的是动态时间伸缩算法(DTW),研究对象是特定人的语音识别,并在MATLAB平台上实现。为了进行后续研究,首先使用电脑中的录音系统录制了阿拉伯数字0—9的语音文件,并转化成 “.wav”格式的文件。-Abstract: This thesis mainly studied the basic principle of speech recognition, to analyze the composition of the speech recognition system, including the preprocessing, feature extraction, to set up the module library, identify several most matches. Pretreatment, including speech sampling, pre-emphasis, add window (hamming window), endpoint detection Feature extraction of MFCC MEL frequency cepstrum coefficient. The voice system USES a dynamic time scale (DTW) algorithm, the research object is the speaker-dependent speech recognition, and realized in MATLAB platform.To carry out the follow-up study, the first to use the recording in a computer system to record the audio files of Arabic Numbers 0-9, and translated into . Wav format file.
Platform: | Size: 9216 | Author: silver teng | Hits:

[Voice Compressyujiazhong

Description: 预加重程序,仿真《通信原理》教材中的例子,对语音进行预加重处理(The pre-emphasis program is used to simulate the example of communication principle, and the speech is pre-emphasis)
Platform: | Size: 273408 | Author: silion | Hits:

[Communication-Mobilepre-emphasis

Description: 语音信号预加重的仿真分析,抑制信号的高频部分。(Simulation of the pre-emphasis on a voice signal, compressing the high-frequency part of a signal.)
Platform: | Size: 547840 | Author: angzhou365 | Hits:

[matlabYCL

Description: 对信号进行梅尔倒谱系数特征采集之前的预处理,包括分帧加窗、基于短时能量的双门限判别、除噪、预加重等(The preprocessing of Mel cepstrum coefficients before the signal is carried out, including frame windowing, double threshold discrimination based on short-time energy, denoising and pre emphasis)
Platform: | Size: 1024 | Author: 明明不到20字节 | Hits:
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