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Description: 一个比较简单的语音识别程序，使用windows Speech API，语音到文本。-One of the simple programs for speech signal recognition. using windows Speech API, speech to text.
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Size: 326656 |
Author: 陈想 |
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Description: LPC美尔倒谱特征的C源程序
这是我根据赵力那本“语音信号处理”后面的源代码写的LPC美尔倒谱特征的C源代码，
基本上没做什么修改，需要的朋友就可以不用自己再录入了。
程序调试通过，可以直接用。
-LPC Cepstrum C source code is under Zhao The "speech signal processing," the source generation behind code written by LPC Cepstrum C source code, is basically what changes need a friend would not have had his own input. Through debugging procedures can be directly used.
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Size: 1024 |
Author: handman |
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Description: 此程序由matlab实现,用于对语音信号的功率谱密度进行估计.希望能有所帮助.-this procedure from Matlab realized, for the speech signal power spectral density estimation. hope to be helpful.
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Size: 3072 |
Author: rw |
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Description: 离散时间语音信号处理源码（matlab）-discrete-time speech signal processing FOSS (Matlab)
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Size: 4211712 |
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Description: 自编的语音语谱图程序，适用于语音信号处理以及语音增强处理，-self Speech Spectrogram procedures applicable to the speech signal processing and speech enhancement.
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Size: 1024 |
Author: tang |
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Description: dsp实时处理，matlab仿真程序，应用于语音信号处理-dsp real-time processing, matlab simulation program, used in speech signal processing
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Size: 58368 |
Author: fengyun |
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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；最后，设计一个信号处理系统界面。-Record a person s own voice signal, and recording the signal sampling draw sample after the speech signal time-domain waveform and frequency spectrum given filter performance indicators, using a window function method and the bilinear transform filter design and draw the frequency response filter then use the filter of their own design for the signal acquisition filtering, after filtering the signal to draw the time-domain waveform and spectrum, and filtering of signals before and after contrast, analysis of signal changes playback of voice signal Finally, the design of a signal processing system interface.
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Size: 2048 |
Author: yim |
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Description: 说明小波消噪的理论，阈值估算和提取，给出在MATLAB下阈值估算和消噪的函数，并在语音信号处理中增强语音。-Explain the theory of wavelet denoising, threshold estimation and extraction, are given in the MATLAB under the threshold estimation and de-noising function, and speech signal processing to enhance voice.
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Size: 69632 |
Author: 宋知用 |
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Description: 语音端点检测代码 用matlab实现短时语音信号 端点检测!希望有用!-Speech Endpoint Detection matlab code used to achieve short-time speech signal endpoint detection! Hope useful!
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Size: 36864 |
Author: hsw0320 |
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Description: 输入自己事先录制好的语音信号，该程序实现将语音信号通过FIR低通、高通、带通滤波器后还原出来的语音质量-Enter their own pre-recorded speech signal, the program will realize voice signal through the FIR low-pass, high pass, band-pass filter to restore them after the voice quality
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Size: 150528 |
Author: shelly |
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Description: 录制一段个人自己的语音信号。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；对语音信号进行加噪和去噪处理，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；实现快录慢放、慢录快放等功能。-Record a person' s own voice signal. Of the recorded signal sampling draw sample after the speech signal time-domain waveform and frequency spectrum of voice signals, noise and de-noising processing, draw Filtered time-domain signal waveform and spectrum, and filtering the signal before and after comparative analysis of signal changes playback voice signal realize quickly recorded slow release, slow release recorded faster functions.
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Size: 243712 |
Author: 或或 |
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Description: 波形和频谱双显示
记录讲话直接进入MATLAB
手动分割讲话波形-创建标签文件
波形编辑-切割，复制或粘贴
共振峰分析-显示共振轨道的F1 ， F2和F3
基音分析
过滤工具-语音信号滤波器截止频率
比较工具-比较两个波形的频谱距离使用几种措施
增加噪声-Dual time-waveform and spectrogram displays
Records speech directly into MATLAB NEW
Displays time-aligned phonetic transcriptions [e.g., TIMIT s .phn files]- see example Figure above
Manual segmentation of speech waveforms- creates label files which can be used to train speech recognition systems
Waveform editing- cutting, copying or pasting speech segments
Formant analysis- displays formant tracks of F1, F2 and F3
Pitch analysis
Filter tool- filters speech signal at cutoff frequencies specified by the user
Comparison tool- compares two waveforms using several spectral distance measures
Speech degradation- adds noise to the speech signal at an SNR specified by the us
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Size: 1180672 |
Author: wsw |
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Description: 语音信号采集与分析
简单地讲，可以在原声音流中叠加延迟一段时间后的声流，实现回声效果。当然通过复杂运算，可以计算各种效应的混响效果。如此产生的回声，我们称之为数字回声。初始化配置： 05 户通过 12C 总线将配置命令发送到 AIC23 ，配置完成后 AIC23 开始工作。语音信号的输入： AIC23 通过其中的 AO 转换采集输入的语音信号，每采集完一个信号后，将数据发送到 05 户的 McBS 户接口上， 05 户可以读取到语音数据，每个数据为 16 位无符号整数，左右通道各有一个数值。语音信号的输出： 05 户可以将语音数据通过 McBS 户接口发送给 AIC23 , AIC23 的 OA 器件将他们变成模拟信号输出。-Speech Signal Acquisition and Analysis
Simply put, you can stream at the original voices superimposed delayed for some time after the acoustic streaming, achieve echo effects. Of course, through complex calculations, can calculate a variety of reverb effects. Echo so produced, we call the digital echo. Initialization, configuration: 05 through 12C bus to configure the command sent to the AIC23, configured to work after the completion of AIC23. Speech Signal input: AIC23 through one of the AO Acquisition conversion of voice input signal, a signal after each acquisition, the data is sent to the 05 interface on McBS households, 05 can be read into the voice data, each data is 16 bit unsigned integer, a value about each channel. Voice signal output: 05 may be voice data sent through the interface McBS families give AIC23, AIC23 of OA devices into their analog signal output.
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Size: 116736 |
Author: fishsky |
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Description: 语音信号处理-对输入语音信号进行端点检测和基音轨迹的检测（利用acf、amdf和acf/amdf三种方法）。-Speech Signal Processing- the input voice signal trajectory endpoint detection and pitch detection (using acf, amdf and acf/amdf three methods).
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Size: 3072 |
Author: 冰 |
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Description: 语音信号处理前的预处理部分，包括预加重，分frame，加窗，是语音信号编程入门的一个很好的参考程序-Speech signal processing part of the pre-pre-treatment, including pre-emphasis, sub-frame, plus window, the speech signal a good entry-programming reference procedures
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Size: 1024 |
Author: 张阳 |
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Description: oppenheim digital signal processing book
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Size: 7462912 |
Author: farzaneh |
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Description: it reconstructs the speech signal using LPC approach
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Size: 245760 |
Author: ramnik |
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Description: 该程序运用NLMS算法对两端语音信号进行分离，模型采用了回波抵消器的数学模型，并附加了处理后的语音信号，效果明显！-The program ends using NLMS algorithm for speech signal separation, model Echo Canceller using a mathematical model and attached processed voice signal, the effect obviously!
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Size: 3036160 |
Author: 成林 |
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Description: 采样小波包分解语音信号，分解为3层，并求出分解系数-Speech signal using wavelet packet decomposition，decomposing 6 level，and attain decomposition coefficient
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Size: 1024 |
Author: 媛媛 |
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Description: 详细说明：语音信号的短时分析，主要包括：分帧、短时能量、短时平均幅度、短时过零率、短时自相关函数、短时幅度差、倒谱、复倒谱、lpc系数、lpc谱估计等 绝对保证质量，是保研后导师布置的一些基础程序-Details: short-time speech signal analysis, including: framing, short-term energy, short-term average rate, short-time zero crossing rate, short-time autocorrelation function, short-term magnitude difference, cepstrum, complex cepstrum, lpc coefficient, lpc absolute guarantee of the quality of spectral estimation, is the security arrangement of some of the research base after the mentor program
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Size: 6144 |
Author: 林溪 |
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