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Description: 在MATLAB中用Goertzel算法(先用FFT算法,再用goertzel算法,)仿真DTMF的产生与检测的程序。应用于双音多频信号的产生与检测。-in Goertzel algorithm using MATLAB (first using FFT algorithm, reuse goertzel algorithm) simulation DTMF detection with the selection process. Used DTMF signal detection with the selection.
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Size: 1024 |
Author: 徐政 |
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Description: 语音识别中的MFCC参数提取的java源码和FFT的java源码-Speech Recognition MFCC parameter extraction of the java source and FFT of the java source code
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Size: 9216 |
Author: LT |
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Description: 在Visual studio 2005下用C#开发的说话人确认(speaker verification)实现。
包括三部分:
1、VoicePrintExistenceCheck - 检查是否存在某用户ID的声音指纹
2、SpeakerVerifyTextDependRegister - 如果不存在,则将其注册到系统中(可以通过语音文件,也可以直接通过麦克风直接输入语音)。
3、SpeakerVerifyTextDependVerify - 确认一个测试样例是否和用户的声音指纹相匹配。
源代码包含如下几个部分:
* speakerVerify.* - 本项目的工程文件(用于测试)
* SpeakerRecognitionActivityLib - 说话人确认的各个步骤的源代码(共12个C#源代码文件)。
* DictationActivitySpeechFX - 使用System.Speech编写的dictation功能(DictationActivity.cs,DictationActivity.Designer.cs)。
* DictationActivitySapi51 - 使用SAPI5.1编写的dictation功能(DictationActivitySapi.cs,DictationActivitySapi.Designer.cs)。
还有其他项目运行所需的源代码和配置文件等。
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Size: 103424 |
Author: 杨小伟 |
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Description: 基于矢量量化和查找表的改进DTW语音识别方法,对算法进行了详细的研究与模拟.-Based on Vector Quantization and Improved look-up table DTW speech recognition method, the algorithm carried out a detailed research and simulation.
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Size: 143360 |
Author: chenfeng |
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Description: ears-0.32, linux下有用的语音信号处理工具包,包括音频文件的读取,fft的计算和critical bands的计算-ears-0.32, linux under the useful speech signal processing toolkit, including the audio files to read, fft calculation and the calculation of critical bands
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Size: 188416 |
Author: liu yu |
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Description: 语音参数短时能量,短时过零率,短时平均幅度,语音端点检测,fft变换与反变换-Voice parameters of short-time energy, short-time zero-crossing rate, short-term average rate, voice activity detection, fft transform and inverse transform
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Size: 3072 |
Author: 路培桄 |
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Description: this program deals with simple speech processing like fft, use of hamming window, my durbin algorithm and likes. this program has gui too.
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Size: 131072 |
Author: switspk |
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Description: DSP, FFT, filters, speech
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Size: 26624 |
Author: Silvie Vyhnankova |
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Description: 自己编写的语音信号的采集,fft变换(两种),以及信噪比的计算!希望对大家有所帮助!-I have written the speech signal acquisition, fft transform (two kinds), and the calculation of signal to noise ratio! We want to help!
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Size: 3912704 |
Author: 祁广杰 |
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Description: 语音特征参数分析,包括语音信号的预处理,FFT变换及逆变换-Speech characteristic parameters analysis, including the speech signal pre-processing, FFT transform and inverse transform
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Size: 46080 |
Author: vinson |
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Description: fft exemple. sistem speech recognition
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Size: 1024 |
Author: raden ozt |
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Description: 观测实时模拟信号(语音)的频谱
用音频设备采集一段语音,将语音存为.wav格式。对wav文件作分段傅里叶变换分析。语音是分音节的,应把它分段分析,而且实际运用中的数字信号处理的FFT的点数是有限的,一般只能达到千点。用傅里叶反变换IFFT,从频域恢复信号。画出频谱图和语音波形图。
-Observing real-time analog signal (voice) of the spectrum collected with the audio devices section of voice, voice saved as. Wav format. Wav file of the sub-Fourier transform analysis to make. Speech is divided into syllables, it should be sub-analysis, and practical application of digital signal processing of the FFT points is limited, generally only reach 1000 points. Fourier inverse transform IFFT, to restore the signal from the frequency domain. Draw the spectrum map and voice waveform.
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Size: 63488 |
Author: benny |
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Description: 信号频谱分析
1. 观测已知方波信号、正弦波信号的频谱
分析频率为4KHz、8KHz的占空比为50 的方波信号、正弦波信号的频谱。记录所得频谱图。
2. 观测实时模拟信号(语音)的频谱
用音频设备采集一段语音,将语音存为.wav格式。对wav文件作分段傅里叶变换分析。语音是分音节的,应把它分段分析,而且实际运用中的数字信号处理的FFT的点数是有限的,一般只能达到千点。用傅里叶反变换IFFT,从频域恢复信号。画出频谱图和语音波形图-Signal spectrum analysis 1. Observing a known square-wave signal, sine wave signal frequency spectrum analysis 4KHz, 8KHz for the 50 duty cycle square wave signal, sine wave signal frequency spectrum. The recorded frequency spectrum. 2. Observing real-time analog signal (voice) of the spectrum collected with the audio devices section of voice, voice saved as. Wav format. Wav file of the sub-Fourier transform analysis to make. Speech is divided into syllables, it should be sub-analysis, and practical application of digital signal processing of the FFT points is limited, generally only reach 1000 points. Fourier inverse transform IFFT, to restore the signal from the frequency domain. Draw maps and voice waveform spectrum
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Size: 64512 |
Author: benny |
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Description: 信号频谱分析
1. 观测已知方波信号、正弦波信号的频谱
分析频率为4KHz、8KHz的占空比为50 的方波信号、正弦波信号的频谱。记录所得频谱图。
观测实时模拟信号(语音)的频谱
用音频设备采集一段语音,将语音存为.wav格式。对wav文件作分段傅里叶变换分析。语音是分音节的,应把它分段分析,而且实际运用中的数字信号处理的FFT的点数是有限的,一般只能达到千点。用傅里叶反变换IFFT,从频域恢复信号。画出频谱图和语音波形图-Signal spectrum analysis 1. Observing a known square-wave signal, sine wave signal frequency spectrum analysis 4KHz, 8KHz for the 50 duty cycle square wave signal, sine wave signal frequency spectrum. The recorded frequency spectrum. Observing real-time analog signal (voice) of the spectrum collected with the audio devices section of voice, voice saved as. Wav format. Wav file of the sub-Fourier transform analysis to make. Speech is divided into syllables, it should be sub-analysis, and practical application of digital signal processing of the FFT points is limited, generally only reach 1000 points. Fourier inverse transform IFFT, to restore the signal from the frequency domain. Draw maps and voice waveform spectrum
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Size: 94208 |
Author: benny |
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Description: This code designed for CCS for speech recogntion using FFT based feature extraction.
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Size: 110592 |
Author: Mahesh |
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Description: 此文档是一份经典算法FFT的演讲PPT,包含最小8点FFT源码。-This document is a classical algorithm for FFT' s speech PPT, with minimum 8 FFT source code.
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Size: 109568 |
Author: 牟思敏 |
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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.
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Size: 13312 |
Author: 赵欣 |
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Description: 语音信号的频谱分析,包括采样语音的FFT变换,及倒频谱分析.-Speech signal spectrum analysis, including FFT, voice samples, and cepstrum analysis.
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Size: 65536 |
Author: leiwi |
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Description: MFCC for speech recognition with java.
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Size: 17408 |
Author: aymen |
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Description: MATLAB features a family of application-specific collections of
functions called toolboxes. These extend the MATLAB environment in order
to solve particular classes of problems. Areas in which toolboxes are available
include signal processing, control systems design, dynamic systems simulation,
systems identification, neural networks, and others.
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Size: 419840 |
Author: khang
|
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