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hmm
基于语音信号工具箱的隐马尔科夫HMM模型的说话人识别，请联系lishicheng64@126.com(Speech signal based on HMM toolbox Hidden Markov Model speaker recognition, please contact lishicheng64@126.com)
- 2010-03-03 14:38:48下载
- 积分：1
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recognition
一种加权K近邻语音情感识别程序，包括部分柏林语音情感库(A Weighting K-nearest neighbor speech emotion recognition program, including some libraries Berlin Speech Emotion)
- 2013-09-04 16:15:24下载
- 积分：1
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AutoPhoneSys
自动语音应答系统
硬件需要一个语音卡或一个语音modem
其中有6个类，下载包里头有详细类说明(Automatic Voice Response System hardware needs a sound card or a voice modem which six categories, Download Packet Erlitou detailed presentation on)
- 2007-01-10 14:21:59下载
- 积分：1
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loudness
计算声音信号的总声压级，1/3/倍频程声压级，总响度，特性响度，尖锐度(Calculation of the total sound pressure level of the voice signal, 1/3/octave sound pressure level, the total loudness, loudness characteristics, acute severe)
- 2021-05-14 14:30:02下载
- 积分：1
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The-FIR-filter-design
（附带有文档，文档中有方案，有程序有结果，设计完后对一段语音进行处理）给定滤波器的规一化性能指标（参考指标,实际中依据每个同学所叠加噪声情况而定）通带截止频率wp=0.25*pi。 采用窗函数法设计低通、高通、带通型FIR滤波器，对叠加噪声前后的语音信号进行滤波处理，绘出滤波器的频域响应，绘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；在相同的性能指标下比较各方法的滤波效果，并从理论上进行分析((With the document, document, program, program results, the design after a voice) to set the filter rules of performance indicators (reference, practice, each student adding noise may be) passband cutoff frequency wp = 0.25* pi. Window function method to design low pass, high pass, band-pass FIR filter, filtering the voice signal before and after adding noise, draw the filter frequency response plot the time-domain waveform and spectrum of the filtered signal, and filtering the signal before and after comparison and analysis of signal changes the filtering effect of the various methods in the same performance indicators, and the theoretical analysis)
- 2021-01-15 00:28:46下载
- 积分：1
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rebuilt.HumanTimbre_recognition-master
利用mfcc特性进行语音识别，分类算法为svm。(Speech recognition using MFCC characteristics)
- 2018-03-31 11:16:48下载
- 积分：1
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DTW
该文件为基于DTW的孤立词识别系统（特定人、小词汇量，未采用鲁棒语音识别技术）(The document DTW for isolated word recognition system based on (a specific person, small vocabulary, did not use robust speech recognition technology))
- 2014-09-23 10:28:25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
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sffddpsolap
语音合成程序源码！psalo频域基音同步叠加方法。它首先对原始语音信号进行短时频域变换，的到短时谱与短时谱包络。短时谱除以短时谱包络的到声源短时谱，对声源短时谱的实部与与虚部分别进行线性插值，就能达到改变变语音信号基频的目的，然后再进行频域反变换，可的到变换后的短时语音信号。短时谱包络部分也能独立改变，以达到改变音色的目的。
(Voice synthesis program source! psalo frequency domain pitch synchronous superposition method. It was first carried out on the original speech signal a short-time frequency-domain transform, to the short-time spectrum and short-time spectrum envelope. The short-time spectrum divided by the short-time spectrum envelope of the short time spectrum of the sound source, the short time spectrum of the real part of the sound source, and the imaginary parts of the linear interpolation, can achieve the purpose of changing the fundamental frequency of the alternating speech signal, and then then the inverse transform of the frequency domain, can be to the short-time speech signal after conversion. The short-time spectral envelope section can be varied independently, in order to achieve the purpose of changing the tone.)
- 2012-09-09 23:36:03下载
- 积分：1
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vqonMatlab
1 训练
打开Matlab 使Current Directory为VQ所在的文件夹(比如：E:vq)
在Command windows中输入
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- 2021-04-20 10:28:50下载
- 积分：1
