资源列表
record
- 這是一個簡單的錄音程式-record
LMS
- LMS滤波器(最小均方误差滤波器),学习现代数字信号处理理论的人应该用的上。
jiafa1
- 连加器运算 , 可以输入任意数值惊醒0—这一数值的连加计算-Even adder operation, you can enter any number awakened 0- the value calculated with added
voicerecognition
- 自己改写的语音识别的代码,matlab实现-voice recognition
变声软件
- 这是基于matlab语言实现的一个GUI操作界面,可通过调节语音信号的频率改变声音的效果(This is a GUI operating interface based on matlab language, which can change the sound effect by adjusting the frequency of speech signals)
dsp
- 本压缩包含解差分方程,z变换,滤波器设计,回响滤波器,采样例子多个dsp的基础知识的相关函数。(Solving difference equation, z transform and filter design)
cochleagram
- cochleagram processing
Beamform-final
- 这是基于波束形成方法的声源定位,一般信号处理用matlab语言,利用C语言进行了修改。(This is the sound source localization based on the beamforming method. The general signal processing is modified by the MATLAB language and is modified by the C language.)
SOUNDC
- 扬声器开发 C 语言源程序库 ,用C语言进行扬声器的研究. -Speakers to develop C language source code library, using C language speakers research.
MATLAB-SINGAL-PROCESSING
- 对采集的语音信号进行滤波,谱分析,并进行前后时域和频域波形比较。可选择IIR和FIR滤波器进行高通,低通和带通滤波。-To obtain the voice signal filtering, spectrum analysis, and the time domain and frequency domain waveform and compared. Can choose IIR and FIR filter for qualcomm, low pass and bandpass filt
iir
- 语音信号的iir滤波器设计,从带有噪音的信号中提取原始声音.目前,MP3播放器一般功率放大器的工作频率范围就是这个范围。但是大部分有用的和可理解的信息的频率在200到3500Hz之间。所以我们可以在这个范围间滤波,达到使声音可理解的要求。现将数字滤波器的设计指标设为通带截止频率fb=600HZ,阻带频率fc=1200HZ,通带波纹Ap=1dB,阻带波纹As=40dB,要求确定H(z)。-design of the iir filter, get the original voice withou
frft
- 基于统计特征的语种识别算法分析与实现。1.提取语音的客观统计特征;2、通过分类器建立训练学习模型;3、将模型运用于汉语、英语、日语等语种识别实验,与人的主观感觉做对比-Based on statistical language identification algorithm analysis and Realization of objective statistics. The extraction of speech features 2, the classifier built th