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GSM_full_rate.RAR
- 长期预测(LTP)与规则脉冲激励(RPE),而全速率编解码器就被称为RPE-LTP线性预测编码器。 输入至RPE-LTP编码器的数据为包括160个采样值的20ms语音,每一个采样值都拥有13位精度。数据首先通过预加重滤波器来提高信号的高频分量,以获得更好的传输效率。滤波器一般还消除信号上的任何偏移以简化进一步的计算。 正如前面所提到的,语音产生模型可看成是空气通过一组不同大小的圆柱体。短期分析级采用自动相关来计算与模型所用的8个圆柱体有关的8个反射系数,同时采用一种称为S
totalchengxu
- lpf滤波 非常无敌超级好用的C或Java源码或其他好源码或编程学习资料-lpf filter is invincible super-easy to use C or Java source code, or other good source code or programming learning materials
BasedonMATLABspeechsignalspectrumanalysisandfilter
- 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-The individual' s own record a voice signal, and the recorded signal is sampled draw sampled
LMS_speech_enhancement
- 运用自适应滤波算法实现的语音增强Matlab程序，附带带噪语音，在M文件中自己修改文件名即可使用-The use of adaptive filtering algorithm for speech enhancement Matlab program, with noisy speech, changes in the M file, the file name to use their own
kalman_speechenhancement
- 卡尔曼滤波程序实现语音增强算法，在Matlab上程序实现了该算法-Program for Kalman filtering speech enhancement algorithm in Matlab program implements the algorithm on
lms
- 使用最小均方算法，对声音进行滤波。去掉声音中混有的噪音。-Using the least mean square algorithm, the sound is filtered. Some of the sound mixing to remove noise.
Speaker_recognition
- 基于mathmatic环境下的语者识别介绍本程序通过滤波、去噪、建模、匹配等过程，将录音人的声音建立成模型，当录音人再次输入声音时能自动识别说话人的身份。-based on mathmatic,that will recognise the speaker
Voice-signal-digital
- 语音信号的数字滤波 fir数字滤波器的海明窗函数设计-Voice signal digital filter fir digital filter design of the Hamming window function
Untitled1
- 带噪语音的lms自适应滤波增强算法实现,用于实现带噪语音的语音增强-Wideband and Sinusoidal Noise Reduction Based on Adaptive Filter with Variable Step Size
Untitled7
- 基于自适应滤波的lms变步长带噪语音的正弦信号仿真程序，假定语音信号为正弦信号-LMS based on adaptive filter variable step of long belt chirp voice frequency sine signals, assuming that the simulation program for sine signals of speech signal
Untitled8
- 基于自适应滤波的固定步长lms带噪语音仿真，假定语音信号的单频正弦信号-Based on adaptive filter fixed step length LMS belt chirp voice simulation, assume speech signal single-rate sine signals
Untitled9
- 基于自适应滤波的nlms带噪语音仿真，假定语音信号是经过给定系统的信号-NLMS based on adaptive filter belt chirp voice simulation, assume speech signal is a after a given system signal
Untitled10
- 基于自适应滤波的rls带噪语音仿真，假定语音信号是一经过给定系统的信号-RLS adaptive filter based on the belt chirp voice simulation, assume speech signal is a after a given system signal
work
- 语音信号的录入和滤波程序，百分之百正确，一定要看啊-Speech signal input and filtering programs, 100 correct Must look
YU-YINXINGHAO
- 本文是根据网上找的“信号的分析与处理综合实验”的内容，通过学习和MATLAB实践后的学习总结。实验内容为真实语音信号的采样重构，具体要求如下：录制一段自己的语音信号，并对录制的信号进行采样；画出采样前后语音信号的时域波形和频谱图；对降采样后的信号进行插值重构，滤波，恢复原信号。-failed to translate
Voice-integer-IIR-filter-C-program
- 语音的IIR滤波程序，可以实现对语音信号的IIR整型滤波-Voice of the IIR filter program, you can achieve integer IIR filtering of speech signals
Kalman
- 基于卡尔曼滤波的语音增强算法研究论文.可以-Speech enhancement based on Kalman filter algorithm
yuyingfenxi
- 录制一段自己的语音信号，时间为10s左右，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；采用窗函数法或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语音信号进行滤波，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.-Record a voice signal their time was 10s or so, and the recorded signal is sampled draw sampled s
voice-reduction
- 高噪声环境下基于自适应滤波语音降噪技术的研究.zip-voice reduction technology research based on adaptive filtering under high noise environment
Wiener
- 经典的weiner滤波程序，用于语音噪声过滤-Classic weiner filtering procedure, the noise filter for voice