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C_ZTQ
基于最小均方误差的助听器语音去噪算法, 用于以最大程度消除助听器接受的语音杂音.(Speech denoising algorithm based on the minimum mean square error of a hearing aid for the hearing aid acceptable voice murmur eliminated to the greatest extent.)
- 2012-10-08 17:20:58下载
- 积分：1
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duoxinhaofenli
多信号分离MUSIC的经典学习程序
对初学者帮助很大(it is a baisc and useful code of the multiple signal classfication for the beginners.)
- 2013-09-11 20:32:20下载
- 积分：1
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The-FIR-filter-design
（附带有文档，文档中有方案，有程序有结果，设计完后对一段语音进行处理）给定滤波器的规一化性能指标（参考指标,实际中依据每个同学所叠加噪声情况而定）通带截止频率wp=0.25*pi。 采用窗函数法设计低通、高通、带通型FIR滤波器，对叠加噪声前后的语音信号进行滤波处理，绘出滤波器的频域响应，绘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；在相同的性能指标下比较各方法的滤波效果，并从理论上进行分析((With the document, document, program, program results, the design after a voice) to set the filter rules of performance indicators (reference, practice, each student adding noise may be) passband cutoff frequency wp = 0.25* pi. Window function method to design low pass, high pass, band-pass FIR filter, filtering the voice signal before and after adding noise, draw the filter frequency response plot the time-domain waveform and spectrum of the filtered signal, and filtering the signal before and after comparison and analysis of signal changes the filtering effect of the various methods in the same performance indicators, and the theoretical analysis)
- 2021-01-15 00:28:46下载
- 积分：1
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GSC
采用广义旁瓣抵消（GSC）自适应波束形成方法实现时域和频域滤波，采用LMS自适应算法，最终实现语音增强。（文件中包含纯净语音及不同信噪比的带噪语音）(Generalized sidelobe canceller (GSC) adaptive beamforming method to achieve time-domain filtering using the LMS adaptive algorithm, and ultimately the speech enhancement)
- 2020-12-28 23:19:02下载
- 积分：1
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jinchangMUSIC2
近场的MUSIC算法，可用于近场噪声源识别(Near-field MUSIC algorithm can be used for near-field noise source identification)
- 2021-04-03 16:49:06下载
- 积分：1
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用Matlab实现DTW孤立词识别
做语音识别时用到DTW 算法。便于初学者的学习。(DTW algorithm is used for speech recognition. Easy for beginners to learn.)
- 2020-12-07 12:39:22下载
- 积分：1
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speech-signal-processing
用matlab对采集到的语音进行语音信号处理 包括设置各种滤波器 低通 高通 以及带通
设置好后 观察结果 (speech signal processing)
- 2012-04-25 09:16:52下载
- 积分：1
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LPCC
线性预测倒谱系数（Linear Prediction Cepstrum Coefficient,LPCC）是线性预测系数（Linear Prediction Coefficient,LPC）在倒谱域中的表示。该特征是基于语音信号为自回归信号的值设，利用线性预测分析获得倒谱系数。(Linear Prediction Cepstrum Coefficient)
- 2018-04-18 16:27:27下载
- 积分：1
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SPECTRUM_LMS
1、文件夹中包含了经典功率谱估计和自适应均衡算法两个实验的所有程序。
2、R.m、LMS.m、LMSmain.m为自适应均衡算法的程序：
R.m用来计算输入信号的自相关矩阵及其特征值；
LMS.m为时域LMS算法，用统计的方法仿真得出不同信道参数和不同步长下的学习曲线； LMSmain.m为实验主程序，按照实验要求中的具体数据得到实验结果和曲线。
3、functionx.m、fzhouqitu.m、spectrum.m、bt.m、bart_lett.m、welch.m、SPECTRUMmain.m为经典谱估计的程序：
functionx.m产生需要进行谱估计的函数；
fzhouqitu.m用来计算信号周期图的函数；
spectrum.m是用周期图法进行谱估计的函数
bt.m是用BT图法进行谱估计的函数
bart_lett.m是用BARTLETT法进行谱估计的函数
welch.m是用WELCH法进行谱估计的函数
SPECTRUMmain.m是主程序，按照实验要求中的具体数据得到实验结果和曲线。 (a document folder contains the classic power spectrum estimation and adaptive equalization algorithm for the two experiments all the procedures. 2, R.m, LMS.m, LMSmain.m adaptive equalization algorithm for the procedure : R.m used to calculate the input signal and the correlation matrix eigenvalue; LMS.m too LMS algorithm, use statistical simulation method come to a different channel parameters and the synchronous learning curve; LMSmain.m main program for the experiment, according to the experimental requirements of the specific data and the curve of the experimental results. 3, functionx.m, fzhouqitu.m, spectrum.m. bt.m, bart_lett.m, welch.m. SPECTRUMmain.m classical spectrum estimation procedures : functionx.m have a need for spectrum estimation function; fzhouqitu.m used to calculate t)
- 2007-04-29 17:54:54下载
- 积分：1
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speechstt
c#写的语音识别程序，识别效率还算可以。且根据识别可以给出相应的识别百分比。适合研究语音识别以及tts的参考！(c# to write the speech recognition process, identify efficiency figure. And recognition can be given according to the percentage of the corresponding identification. Tts speech recognition, as well as for research reference!)
- 2010-06-01 09:51:14下载
- 积分：1
