Description: LMS滤波器示例程序，在TURBOC中运行
这是一个简单的可图形显示的C程序
输入信号是一个被噪声污染了的sin信号。 */
/* 运行后，屏幕的上方是输入信号，下方是经过LMS滤波后的输出信号
-LMS filter sample programs, run in TurboC This is a simple graphical display can process the C input signal is a noise contaminated signal sin.*//* Running, the top of the screen is the input signal, the bottom is the result after the LMS filter output signal Platform: |
Size: 4096 |
Author:蜗牛 |
Hits:
Description: 对原始语音进行上采样和下采样，分析对比他们的时域图和频谱图。最后对比原始语音、75HZ激励、150HZ激励以及噪声激励下的效果-SOME SIMPLE MANIPULATIONS OF SOUND USING DIGITAL SIGNAL PROCESSING
The original sound and its spectrogram
Downsampling the waveform
downsampling
Upsampling the waveform
Linear filtering the waveform
Original speech
The source-filter model of speech
Speech with 75-Hz excitation
Speech with 150-Hz excitation
Speech with noise excitation Platform: |
Size: 50176 |
Author:coco |
Hits:
Description: 录制一段个人自己的语音信号，并对录制的信号进行采样；画出采样后语音信号的时域波形和频谱图；给定滤波器的性能指标，采用窗函数法和双线性变换设计滤波器，并画出滤波器的频率响应；然后用自己设计的滤波器对采集的信号进行滤波，画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；回放语音信号-The individual' s own record a voice signal, and the recorded signal is sampled draw sampled speech signal time-domain waveform and frequency spectrum filter performance given by the window function method and bilinear transformation design a filter and draw the filter frequency response then use their own filters designed to filter the collected signals, to draw the filtered time domain waveform and frequency spectrum, and comparing the signal before and after filtering, analysis of signal changes playback of the speech signal Platform: |
Size: 12288 |
Author:姚湘陵 |
Hits:
Description: straight是一个高质量的语音识别、转换、合成工具，构建于matlab编程工具。-STRAIGHT[1] is a high-quality speech analysis, modification and synthesis system based on a simple channel
VOCODER concept (in other words a source filter model). Platform: |
Size: 1844224 |
Author:Hu Feiyue |
Hits:
Description: 近几年来，在自然语言处理、信号处理、随机过程处理等方法的推动下，语音分析与合成技术获得了很大的发展，突破了传统的单纯语音计算算法的研究。情感语音分析与合成的研究，适应了语音技术的未来发展趋势,由于它能够很好的将语音的口语分析、情感分析与计算机技术有机的融合，为实现以人为本，具有个性化特征的语音合成系统，奠定基础。-STRAIGHT[1] is a high-quality speech analysis, modification and synthesis system based on a simple channel
VOCODER concept (in other words a source filter model). Platform: |
Size: 82944 |
Author:Hu Feiyue |
Hits:
Description: tpdsola algorithm,Synthesized speech can be produced by several different methods. All of these have some benefits and deficiencies that are discussed in this and previous chapters. The methods are usually classified into three groups:
Articulatory synthesis, which attempts to model the human speech production system directly.
Formant synthesis, which models the pole frequencies of speech signal or transfer function of vocal tract based on source-filter-model.
Concatenative synthesis, which uses different length prerecorded samples derived from natural speech.
The formant and concatenative methods are the most commonly used in present synthesis systems. The formant synthesis was dominant for long time, but today the concatenative method is becoming more and more popular. The articulatory method is still too complicated for high quality implementations, but may arise as a potential method in the future.-tpdsola algorithm,Synthesized speech can be produced by several different methods. All of these have some benefits and deficiencies that are discussed in this and previous chapters. The methods are usually classified into three groups:
Articulatory synthesis, which attempts to model the human speech production system directly.
Formant synthesis, which models the pole frequencies of speech signal or transfer function of vocal tract based on source-filter-model.
Concatenative synthesis, which uses different length prerecorded samples derived from natural speech.
The formant and concatenative methods are the most commonly used in present synthesis systems. The formant synthesis was dominant for long time, but today the concatenative method is becoming more and more popular. The articulatory method is still too complicated for high quality implementations, but may arise as a potential method in the future. Platform: |
Size: 776192 |
Author:cristianviza |
Hits:
Description: 本程序源码界面实现的是在MATLAB下的语音信号处理理，使用是巴特沃斯低通滤波器 可直接使用。 已通过测试。
-The source interface of the program implemented under the MATLAB speech signal processing management, the use of a Butterworth low-pass filter can be used directly. Has been tested. Platform: |
Size: 4096 |
Author:一群舰队 |
Hits:
Description: 最好的语音合成代码，出自日本，是以源-滤波器模型合成语音质量最好的开源代码，注意，此MATLAB代码，只有P-CODE 如要源码，需向作者申请-The best speech synthesis code, from Japan, is source- filter model synthesizer voice quality the best open source code, note that this MATLAB code, only the P-CODE to source required to apply to the author Platform: |
Size: 573440 |
Author:shone |
Hits:
Description: 三种语音增强算法的源程序，可以播放处理前后的语音，谱相减法，滤波器法及最小均方误差法-Three speech enhancement algorithm of the source can be played before and after voice spectral subtraction filter method and the minimum mean square error method Platform: |
Size: 98304 |
Author:李羽 |
Hits:
Description: This document makes an introduction to speech communication, Acoustic Theory of Speech: The Source-Filter Model,Speech Models and Features,
Linear Prediction Model of Speech,Harmonic plus Noise Model of Speech, etc. Platform: |
Size: 516096 |
Author:prg |
Hits:
Description: 这是我的adsp的project 维纳滤波在语音增强上的应用，是基于维纳滤波的语音增强的matlab实现，其中包括matlab源码，word文档，以及PPT。并设计了简单的
GUI 对语音信号进行加噪处理，然后再其进行维纳滤波。通过gui可以实时改变加入噪声的参数和维纳滤波器的参数，进行分析。并
有频谱显示和语谱对比显示。-This is my project adsp the Wiener filter applied on speech enhancement is based on the Wiener filter speech enhancement matlab implementation, including matlab source code, word documents, and PPT. And designed a simple GUI for adding noise speech signal processing, then it Wiener filtering. You can join the real-time changes in the parameters and noise parameters Wiener filter through gui, for analysis. And a spectrum display and spectral contrast display language. Platform: |
Size: 766976 |
Author:小毛孩 |
Hits:
Description: 以MATLAB软件为工具，在GUI图形用户界面下针对不同特点的语音信号进行八种不同模式的滤波处理的语音信号处理系统，涉及基于巴特沃思滤波器的IIR滤波器和汉明窗设计的FIR滤波器，能够实现语音文件的打开及自定义路径的存储功能，同时可以实现语音信号加噪和音频倒放功能。-The source code, by means of MATLAB, under the Graphical User Interface, achieves a kind of speech signal processing system with eight different modes of speech signal filters working according to different characteristics of speech signal. The system involves IIR filter based on Butterworth analog filter and FIR filter based on Hamming Window, is able to open and save audio files in the user-defined place and realizes functions of noise addition and audio back-play. Platform: |
Size: 19456 |
Author:李小 |
Hits:
Description: 完整的基于HMM的语音识别系统，最终的权值矩阵就是滤波器的系数，包括最后计算压缩图像的峰值信噪比和压缩效果的源码。- Complete HMM-based speech recognition system, The final weight matrix is ??the filter coefficient, Including the final calculation of the compressed image peak signal to noise ratio and compression of the source. Platform: |
Size: 5120 |
Author:朱冬跃 |
Hits:
Search - source filter speech