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vadsohn
VAD去噪算法，可以有效的过滤静音和无效的语音，同时消除了一些背景噪声。(VAD denoising algorithm, can effectively filter invalid mute and voice, while eliminating some background noise.)
- 2013-10-21 13:38:36下载
- 积分：1
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espeak_1.36
espeak语音生成库，内附有移植使用文档《espeak移植使用》(espeak speech generation library, containing transplanted using the document espeak transplant use )
- 2021-01-03 18:08:56下载
- 积分：1
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fft2hps
语音处理的实用代码，完成耳语音的声韵分割功能(Practical speech processing code, complete the ear speech sound segmentation function)
- 2010-06-19 00:12:10下载
- 积分：1
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MMSE_IC
基于LMS(最小均方误差算法)的自适应滤波 基于LMS(最小均方误差算法)的自适应滤波(based on the LMS (MMSE) algorithm based on the LMS adaptive filtering (minimum mean square error algorithm) the adaptive filtering based on the LMS (minimum mean square error algorithm) Adaptive Filter)
- 2020-06-29 23:00:01下载
- 积分：1
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MFCC
在MATLAB中完成对语音的梅尔倒谱系数的提取(Extraction of Mel cepstrum coefficients)
- 2017-10-23 21:40:39下载
- 积分：1
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speaker-recognition-hmm
hmm 说话人识别 matlab 算法，c++算法，(speaker recognition hmm)
- 2013-05-15 13:38:19下载
- 积分：1
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lunwen
本文主要研究普通计算环境下语音自动识别(ASR)技术的软件实时实现，基于语音识别的基本原理和过程。在VC环境下建立了一套语音识别系统及其演示软件。实验表明，该系统对特定人情况有较高的识别率和一定的实时性(In this paper, under the computing environment on general automatic speech recognition (ASR) technology to achieve real-time software, speech recognition based on the basic principles and processes. VC environment in the establishment of a voice recognition system and its presentation software. The experiments show that the systems are in particular have a high recognition rate and a certain degree of real-time)
- 2009-06-06 15:14:09下载
- 积分：1
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matlab
kalman语音处理的源代码，用处语音增强的处理。(the kalman voice processing source code)
- 2013-04-02 16:51:36下载
- 积分：1
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ZXG
一个对有噪声的音频信号进行自相关滤波的c程序，用的turbo c 3.0工具(One pair of noisy audio signal autocorrelation filtering c program using turbo c 3.0 Tools)
- 2021-03-01 22:29:34下载
- 积分：1
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AnalogVoiceSignal
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.)
- 2010-01-27 16:01:16下载
- 积分：1
