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[Speech/Voice recognition/combinefdpsola

Description: 语音合成程序!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,得到短时谱和短时谱包络。短时谱除以短时谱包络得到声源短时谱,对声源短时谱的实部和虚部分别进行线性插值,就可以达到改变语音信号基频的目的,然后再进行频域反变换,可得到变换后的短时语音信号。短时谱包络部分也可以独立改变,以达到改变音色的目的。-speech synthesis procedures! Psalo frequency domain pitch synchronous superposition method. It is first of the original speech signal for short-time frequency domain transform, to be short-term and short-term spectral envelope spectrum. Short-term spectrum divided by the short-time spectral envelope of the sound source to be short-term spectrum of the sound spectrum of short-term real and imaginary parts of linear interpolation, they could change the voice-frequency signals to the end, then we will proceed to frequency-domain transform, Transform available after the short speech signal. Short-term spectral envelope can be independent of changes to achieve the objective of changing colors.
Platform: | Size: 2553 | Author: hzh | Hits:

[Speech/Voice recognition/combinefdpsola

Description: 语音合成程序!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,得到短时谱和短时谱包络。短时谱除以短时谱包络得到声源短时谱,对声源短时谱的实部和虚部分别进行线性插值,就可以达到改变语音信号基频的目的,然后再进行频域反变换,可得到变换后的短时语音信号。短时谱包络部分也可以独立改变,以达到改变音色的目的。-speech synthesis procedures! Psalo frequency domain pitch synchronous superposition method. It is first of the original speech signal for short-time frequency domain transform, to be short-term and short-term spectral envelope spectrum. Short-term spectrum divided by the short-time spectral envelope of the sound source to be short-term spectrum of the sound spectrum of short-term real and imaginary parts of linear interpolation, they could change the voice-frequency signals to the end, then we will proceed to frequency-domain transform, Transform available after the short speech signal. Short-term spectral envelope can be independent of changes to achieve the objective of changing colors.
Platform: | Size: 2048 | Author: hzh | Hits:

[Speech/Voice recognition/combine111

Description: 本源码可实现基本的语音信号抽样和与插值。-The source can be realized basic voice signal sampling and interpolation.
Platform: | Size: 30720 | Author: liuyu | Hits:

[Speech/Voice recognition/combinematlab_reduce_noise

Description: matlab去除50hz噪声。 我用电脑录了一段声音,里面有50hz的周期噪声(因为受交流电干扰)。而我自己的声音频率最低是90hz。我使用了一个10阶butterworth高通滤波器,边带是70hz(介于50跟90之间)。 问题是,这不能直接用。因为声音文件的采样率是22k,70相对于22k来说太小了。所以我得先把我的声音欠采样,然后再滤波,然后再插值。-matlab remove 50hz noise. I used the computer recorded a voice, there are 50hz cycle noise (due to AC interference). I own voice is the lowest frequency of 90hz. I use a 10-order Butterworth high-pass filter, edge belt is 70hz (the range between 50 to 90). The problem is that this can not be directly used. Because the sound files of the sampling rate is 22k, 70 compared with the 22k run too small. So I have to put my voice due to sampling, and then filtering, and then interpolation.
Platform: | Size: 6144 | Author: 张文斌 | Hits:

[OtherMATLAB1

Description: 声音信号的读入 抽取 及插值 (内附语音信号wav文件)-Sound signals read into the extraction and interpolation (with voice signals wav file)
Platform: | Size: 59392 | Author: zhoud | Hits:

[Speech/Voice recognition/combineCwave

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 | Hits:

[GUI DevelopCwave

Description: 音频文件处理程序。DOS界面,功能包括打开所给音频文件,显示文件信息,并可对该文件进行采样,插值等,从而改变语音质量。-Audio file processing. DOS interface, features include open the given audio file, display file information, the file can be sampled, interpolation, so as to change the voice quality.
Platform: | Size: 3173376 | Author: river | Hits:

[Speech/Voice recognition/combinewave

Description: C语言编写。对语音(.wav文件)做读取,写入操作,并可对文件进行采样抽取和内插-C language. Voice (. Wav files) to do read, write, and samples of documents collected and interpolation
Platform: | Size: 2099200 | Author: river | Hits:

[Speech/Voice recognition/combinevoice-maker

Description: 语音信号的内插与抽取 某大学c++的实践作业-Interpolation of speech signals collected by a university with the practical operation c++
Platform: | Size: 1006592 | Author: 张开来 | Hits:

[Windows Developsffddpsolap

Description: 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.
Platform: | Size: 2048 | Author: moderate | Hits:

[Waveletasmwork

Description: 请封装一个类CWave, 并编写一个软件,可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。 -Please package a class CWave, and write a software, to be completed by a voice signal interpolation and decimation. Wav format, mono storage, encoding PCM voice data. The functions that can be completed as follows: (1) read wav file (2) write wav file (3) voice data interpolation and decimation.
Platform: | Size: 508928 | Author: 钟自豪 | Hits:

[OtherSpeech-signal-processing-source-code

Description: 对语音信号进行采样率的转换,如整数倍的内插和抽取,还可以进行非整数倍的采样率转换。-The sampling rate of the voice signal conversion, such as interpolation and decimation integer multiple of, can also be a non-integer multiple of the sampling rate conversion.
Platform: | Size: 5120 | Author: 张辉 | Hits:

[Other9MATLABCHULIXIN

Description: 第9章共振峰的估算方法259 9.1预加重和端点检测259 9.1.1预加重259 9.1.2端点检测260 9.2倒谱法对共振峰的估算260 9.2.1倒谱法共振峰估算的原理260 9.2.2倒谱法共振峰估算的MATLAB程序261 9.3LPC法对共振峰的估算262 9.3.1LPC法共振峰估算的原理262 9.3.2LPC内插法共振峰的估算263 9.3.3LPC求根法共振峰的估算266 9.4连续语音LPC法共振峰的检测268 9.4.1简单LPC共振峰检测268 9.4.2改进的LPC共振峰检测270 9.5基于HilbertHuang变换(HHT)的共振峰检测274 9.5.1希尔伯特变换275 9.5.2语音信号的另一种模型——AMFM模型278 9.5.3对AMFM模型的分析279 9.5.4语音信号共振峰特征参数提取的HHT方法279 9.5.5基于HilbertHuang变换的共振峰检测步骤和MATLAB程序280-Estimation Chapter 9 259 9.1 formant pre-emphasis and pre-emphasis endpoint detection 259 259 9.1.1 9.1.2 9.2 endpoint detection principle cepstrum estimate 260 to 260 9.2.1 formant formant estimation cepstrum 260 9.2.2 cepstrum estimate estimate estimate formant MATLAB program 261 9.3LPC method formant 262 9.3.1LPC law principle of formant estimation interpolation within 262 9.3.2LPC formants 263 9.3.3LPC Root Law resonance estimate peak 266 9.4 Continuous Speech LPC formant detection method is simple LPC formant 268 9.4.1 268 9.4.2 Improved detection LPC 270 9.5 formant detection based HilbertHuang transform (HHT) 274 9.5.1 formant detection Hill Another model 275 9.5.2 Hilbert transform voice signals- AMFM model 278 9.5.3 Analysis of the AMFM model HHT Method 279 9.5.4 formant speech signal feature extraction based HilbertHuang transform 279 9.5.5 Resonance peak detection step and MATLAB program 280
Platform: | Size: 10240 | Author: 孟稳 | Hits:

[MiddleWarekook.py

Description: 用于android 源码编译/以及修改一些辅助功能,语音,摄像头插值,默认时间,等等,支持ui界面-Android source code for compilation/and modify some auxiliary functions, voice, camera interpolation, the default time, etc., to support ui interface
Platform: | Size: 32768 | Author: 何航军 | Hits:

[matlabSignal-recognition

Description: 函数可以作出时域波形图和频域频谱图,并且计算基因频率自动判断音频文件是男声还是女声声音。ds函数输入变量是文件名和降采样的间隔,通过插值的办法保证了原信号的长度,可以画出时域及频域图像,并且判断男女声,最后播放降采样之后的声音-The function can make time-domain waveform and frequency spectrum, and calculate the gene frequency automatically determine the audio file is male or female voice.Ds function of the input variable is interval file name and down sampling, by interpolation approach to ensure that the original signal length, can draw the time domain and frequency domain of image and sound judgment, men and women, last played down after sampling the sound
Platform: | Size: 2048 | Author: dehenbo | Hits:

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