Description: 语音识别(Speech Recognition)是让机器通过识别和理解过程把语音信号转变为相应的文本或命令的技术。在课题中,通过采用DTW(Dynamic time warping, 动态时间伸缩)算法,对实现孤立词的识别进行了初步探讨和研究,实现了在MATLAB软件环境下孤立词语的语音识别,并针对DTW的主要特点及不足做出了总结。-Speech Recognition (Speech Recognition) machines is through recognition and understanding of a process which put the voice signal into the corresponding text or command technology. The issue, through the use DTW (Dynamic time warping, dynamic time telescopic) algorithm to achieve an isolated word recognition conducted a preliminary exploration and study, the realization of the MATLAB software environment isolated words of Speech Recognition and DTW against the main features and deficiencies make summarized. Platform: |
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Author:序号 |
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Description: 对采样过后的语音信号进行分帧处理,并将分帧后的信号存储为一个矩阵-right after sampling the voice signal processing sub-frame and sub-frame of a signal to the storage matrix Platform: |
Size: 1024 |
Author:宋佳 |
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Description: 自己写的提取音频信号mfcc参数的算法,用matlab实现,欢迎大家参考-Write their own audio signal MFCC extraction algorithm parameters, using matlab to achieve, welcome to the reference Platform: |
Size: 3072 |
Author:zhaodapeng |
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Description: 提取语音信号中mfcc参数,可以用来语音识别-Extraction of Speech Signal in the MFCC parameters, can be used to speech recognition Platform: |
Size: 1024 |
Author:吴杰 |
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Description: In this project we have processed the speech signal with the help of the DIGITAL SIGNAL PROCESSING techniques. The speech signal is given as the input will be verified using speech recognition technique using matlab. We have used Mel Frequency Cepstral Coefficient (MFCC) along with Vector Quantization (VQLBG) and Euclidean Distance to identify different characters. Based on the results, data was send to Parallel Printer Port of the computer & using relay different devices will be controlled.
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Size: 2048 |
Author:SimonKap22 |
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Description: 关于语音处理的英文书籍,其中特征提取部分(MFCC)讲解的很好很详细-The performance of speech recognition systems receiving speech that has been transmitted over mobile channels can be
significantly degraded when compared to using an unmodified signal. The degradations are as a result of both the low
bit rate speech coding and channel transmission errors. A Distributed Speech Recognition (DSR) system overcomes
these problems by eliminating the speech channel and instead using an error protected data channel to send a
parameterized representation of the speech, which is suitable for recognition. The processing is distributed between the
terminal and the network. The terminal performs the feature parameter extraction, or the front-end of the speech
recognition system. These features are transmitted over a data channel to a remote "back-end" recognizer. The end result
is that the transmission channel does not affect the recognition system performance and channel invariability is achieved. Platform: |
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Author:gqy |
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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 |
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Description: 1.用matlab实现对一个元音和一个清辅音的LPC系数,LPCC系数和MFCC系数的提取。
2,用LPC对这两个语音信号的功率谱进行估计,其中,LPC系数要分别有5阶,10阶,15阶和20阶四种情况,并在一个图里画出信号本身的FFT功率谱和四条不同阶数的LPC谱估计图,以作比较。注意,LPC和LPCC只考察自相关法。
3,用对数面积比的方法来求出这两个发音的变截面声管模拟图形,10阶即可-Using matlab to achieve the LPC coefficients of a vowel and a voiceless consonant, the extraction of LPCC Features coefficients and MFCC coefficients. LPC both the speech signal power spectrum estimate, which, LPC coefficients, respectively, five bands, 10 bands, 15 bands and 20 the first four cases, and draw a graph of the FFT power of the signal itself spectrum and four different orders of LPC spectral estimation map, for comparison. LPC and LPCC only examine the autocorrelation method. 3, the number of area ratio method to calculate the variable cross-section of the two pronunciation sound tube simulation graphics, 10 bands can be Platform: |
Size: 72704 |
Author:houjam |
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Description: how to scovert speech to matrix
how to convert matrix form to speech
how to find sample frequency for speech
how to play an sound signal in matlab
mf-how to scovert speech to matrix
how to convert matrix form to speech
how to find sample frequency for speech
how to play an sound signal in matlab
mfcc Platform: |
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Author:nvdmahesh |
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Description: MFCC MATLAB 实现,主要是在分布式语音信号参数提取方面的代码,提取出的参数直接量化编码-MFCC MATLAB implementation, the main voice signal in the distributed parameter extraction of the code, the extracted coding parameters to directly quantify Platform: |
Size: 1024 |
Author:赵新 |
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Description: matlab程序提取语音信号的基音频率、共振峰、能量、MFCC等特征参数。-The matlab program to extract the speech signal fundamental frequency, Gong Zhenfeng, energy, MFCC and other characteristic parameters. Platform: |
Size: 5120 |
Author:吕长勇 |
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Description: 本程序是基于matlab的程序,功能是实现语音信号的mf-This procedure is based on the matlab program, the function is to realize the voice signal MFCC Platform: |
Size: 1024 |
Author:林玉梅 |
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