Description: 实现语音的采集和回放，先将语音信号数字化再存储在ROM中，最后从读出、数模转换、滤波得到模拟语音信号!-achieve Audio Capture and Playback, the first digital voice signal then stored in ROM, the read out from the final, digital-to-analog converter, filter was simulated voice signals. Platform: |
Size: 1871 |
Author:左富强 |
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Description: 这是语音信号中针对音频文件的读取，特别是对于文件头的读取，经过变化可转变成针对音频文件的转化-This the voice signal regarding the audio files to read, especially for the first document read, After changes can be transformed into audio files against the conversion! ! Platform: |
Size: 59546 |
Author:彭涛 |
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Description: 读取语音信号（用matlab的wavread指令），把语音信号分帧、加窗，进行清浊分割，提取基
频，这一部分较简单，自己编程序做。参考文献自己到图书馆期刊网上查找。
提取语音信号的lpc参数，可调用lpcfit.m 程序（我提供，见附件），将源、目标语音的浊音
段的lpc系数进行DTW规整，调用pathita2.m 程序（我提供，见附件）。将规整得到的lpc系数
转换成lsp参数，调用lpcar2ls.m 程序（我提供，见附件）， 再进行转换映射，调用matlab
的指令newrbe。-read the speech signal (using Matlab wavread Directive), Voice Signal frames, Windowed, Qingzhuo segmentation, extracting fundamental frequency, this part is relatively simple, they programmed to do. References their online journals to the library search. Voice Signal Extraction lpc parameters can be called lpcfit.m (I provide, see Annex), the source, Voiced objective voice of the lpc coefficient DTW structured, Call pathita2.m (I provide, see Annex). Structured to the lpc lsp conversion coefficient parameters, calling lpcar2ls.m (I offer see Annex), the shift mapping, called the directive newrbe Matlab. Platform: |
Size: 3928 |
Author:韦敏 |
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Description: 在linux平台下,纯c写的盲信号分离的代码.它采用基于卷积混合的盲信号分离算法,不但可以分离人工合成的混合信号,而且对于真实环境中的卷积混合的语音信号也能够分离.在本程序中,包含了两个测试文件,makefile后便可以直接使用.
另外值得一提的是,压缩包里包含有一些语音处理方面的常用函数.例如fft变换,读取\写入wav文件,以及一些常用的一维向量和二维矩阵变换的函数.这些可以直接应用在其他应用程序里去.-in linux platform, net write c Blind Signal Separation code. it is based on convolution of mixed Blind Signal Separation algorithm, not only can be separated from the artificial synthesis of mixed-signal, but for the real environment convolution of mixed voice signals can be separated. in this procedure, the tests included two documents, makefile can be used directly after. Also worth mentioning is that Compression bag containing some of voice processing functions commonly used. fft transform, for example, to read \ write wav file, and some of the commonly used one-dimensional and two-dimensional vector matrix transformation functions. these can be directly applied to other applications going. Platform: |
Size: 141312 |
Author: |
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Description: 实现语音的采集和回放，先将语音信号数字化再存储在ROM中，最后从读出、数模转换、滤波得到模拟语音信号!-achieve Audio Capture and Playback, the first digital voice signal then stored in ROM, the read out from the final, digital-to-analog converter, filter was simulated voice signals. Platform: |
Size: 2048 |
Author:左富强 |
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Description: 这是语音信号中针对音频文件的读取，特别是对于文件头的读取，经过变化可转变成针对音频文件的转化-This the voice signal regarding the audio files to read, especially for the first document read, After changes can be transformed into audio files against the conversion! ! Platform: |
Size: 59392 |
Author:彭涛 |
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Description: 读取语音信号（用matlab的wavread指令），把语音信号分帧、加窗，进行清浊分割，提取基
频，这一部分较简单，自己编程序做。参考文献自己到图书馆期刊网上查找。
提取语音信号的lpc参数，可调用lpcfit.m 程序（我提供，见附件），将源、目标语音的浊音
段的lpc系数进行DTW规整，调用pathita2.m 程序（我提供，见附件）。将规整得到的lpc系数
转换成lsp参数，调用lpcar2ls.m 程序（我提供，见附件）， 再进行转换映射，调用matlab
的指令newrbe。-read the speech signal (using Matlab wavread Directive), Voice Signal frames, Windowed, Qingzhuo segmentation, extracting fundamental frequency, this part is relatively simple, they programmed to do. References their online journals to the library search. Voice Signal Extraction lpc parameters can be called lpcfit.m (I provide, see Annex), the source, Voiced objective voice of the lpc coefficient DTW structured, Call pathita2.m (I provide, see Annex). Structured to the lpc lsp conversion coefficient parameters, calling lpcar2ls.m (I offer see Annex), the shift mapping, called the directive newrbe Matlab. Platform: |
Size: 4096 |
Author:韦敏 |
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Description: 在噪声环境下语音信号的增强
语音信号为读入的声音文件
噪声为正态随机噪声-In the noisy environment of the enhanced voice signal voice signal for the read noise of the sound files for the normal random noise Platform: |
Size: 1024 |
Author:熊敏 |
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Description: 对语音信号进行53双正交样条小波变换，读入的为语音信号文件-Voice signals on 53 pairs of orthogonal spline wavelet transform, read the documentation for the voice signal Platform: |
Size: 195584 |
Author:王渤帆 |
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Description: 一个用C语言程序实现读写语音文件的语音信号低通滤波例子-A C language program used to read and write audio files of voice signal low-pass filter example Platform: |
Size: 1024 |
Author:董飞 |
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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. Platform: |
Size: 116736 |
Author:fishsky |
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Description: 语音信号处理的最基本的Maylab处理程序，包括读入语音波形，清音浊音的检测，加窗，过零率，短时能量，基音最大值。最后有test给出演示-Speech Signal Processing Maylab the most basic treatment procedures, including read into the voice waveform, voiceless voiced detection, add windows, zero-crossing rate, short-term energy, Maximum pitch. Finally there is give demo test Platform: |
Size: 444416 |
Author:张路 |
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Description: 实现小波语音去噪，基本实现了去除噪音的功能。主要读取信号，分解，去噪，重构。-The realization of wavelet denoising voice, the basic realization of the noise removal function. Read the main signal, decomposition, denoising, reconstruction. Platform: |
Size: 13312 |
Author:王磊磊 |
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Description: 此为用DSP来实现对语音信号的录放程序。希望读大家有用-This is achieved with the DSP voice signal playback program. We would like to read useful Platform: |
Size: 6144 |
Author:张芸 |
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Description: 用C++编写的一个软件，完成对一个语音信号的内插和抽取。语音数据以wav格式、单声道存储，编码方式为PCM。可完成的功能为：
（1）读wav文件；
（2）写wav文件；
（3）对语音数据进行内插和抽取。
-Written in C++, a software, to complete a voice signal decimation and interpolation. Wav format, voice data, single-channel storage, encoding for the PCM. The function to be completed: (1) read wav files (2) to write wav file (3)complete decimation and interpolation to the voice data. Platform: |
Size: 1024 |
Author:梦游 |
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Description: 3． 用VC编程浮点程序实现对语音信号的按帧滤波。
1） 在主程序中读取FIR DF系数文件。
2） 在主程序中按帧读取语音样点文件，每帧180点。
3） 设计浮点滤波子程序，供主程序调用。
4） 保存滤波结果数据到文件中。
5） 用cooledit试听滤波后的语音信号。
-3. VC programming with floating-point program to realize the speech signal by frame filtering. 1) In the main program file to read FIR DF coefficient. 2) The main program reads the voice samples according to the frame files, 180 points per frame. 3) floating-point filter design routines for the main program calls. 4) save the filtered data to the file. 5) Listen with cooledit voice signal after filtering. Platform: |
Size: 2813952 |
Author:陈永尧 |
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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: 利用定时器触发ADC采集语音信号，存储到FLASH中，读回后用SPWM输出，经过低通滤波后可以还原出原始信号波形，效果还不错-Use a timer to trigger the ADC capture the voice signal stored in the FLASH, read back with SPWM output after low-pass filtering can restore the original signal waveform, the results were good Platform: |
Size: 1483776 |
Author:亮 |
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Description: 读取语音信号，提取轻音与浊音的频谱，使用不同窗函数-Read voice signal, to extract the frequency spectrum of the light tone voiced, and the use of different window functions Platform: |
Size: 1024 |
Author:sylar |
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Search - read voice signal