Description: This paper deals with the problem of speech enhancement when a
corrupted speech signal with an additive colored noise is the only
information available for processing. Kalman filtering is known as
an effective speech enhancement technique, in which speech signal
is usually modeled as autoregressive (AR) process and represented
in the state-space domain.-This paper deals with the problem of speech enhancement when a corrupted speech signal wit h an additive colored noise is the only informat ion available for processing. Kalman filterin g is known as an effective speech enhancement te chnique. in which speech signal is usually modeled as aut oregressive (AR) process and represented in th e state-space domain. Platform: |
Size: 102400 |
Author:rifer |
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Description: 1．产生一个连续信号，包含低频，中频，高频分量，对其进行采样，进行频谱分析，分别设计三种高通，
低通，带通滤波器对信号进行滤波处理，观察滤波后信号的频谱。
2.采集一段含有噪音的语音信号（可以录制含有噪音的信号，或者录制语音后再加进噪音信号），对其进行
采样和频谱分析，根据分析结果设计出一合适的滤波器滤除噪音信号-1. Produce a continuous signal, including low-frequency, medium frequency, high-frequency components, its sampling, spectral analysis, are designed for three high-pass, low pass, band-pass filter the signal filtering treatment to observe the signal spectrum after filtering. 2. Acquisition section of the speech signal contains noise (can record a signal containing noise, or voice recording, additional noise after the signal), its sampling and spectral analysis, based on an analysis of the results to design a suitable filter to filter out one noise signal Platform: |
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Author: |
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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 |
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Description: 这是本人编写的一个利用维纳滤波实现语音增强的程序.在输入语音信噪比不是很低的情况下,效果不错.-This is one I prepared realize the use of Wiener filter speech enhancement procedure. In the input voice signal to noise ratio is not very low circumstances, good results. Platform: |
Size: 1024 |
Author:李茉 |
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Description: MATLAB代码,利用卡尔曼滤波实现加入白色噪声后的语音信号的增强.效果不错.-MATLAB code, the use of Kalman filter to achieve by adding white noise to enhance the voice signals. Good results. Platform: |
Size: 1024 |
Author:李茉 |
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Description: 波形和频谱双显示
记录讲话直接进入MATLAB
手动分割讲话波形-创建标签文件
波形编辑-切割，复制或粘贴
共振峰分析-显示共振轨道的F1 ， F2和F3
基音分析
过滤工具-语音信号滤波器截止频率
比较工具-比较两个波形的频谱距离使用几种措施
增加噪声-Dual time-waveform and spectrogram displays
Records speech directly into MATLAB NEW
Displays time-aligned phonetic transcriptions [e.g., TIMIT s .phn files]- see example Figure above
Manual segmentation of speech waveforms- creates label files which can be used to train speech recognition systems
Waveform editing- cutting, copying or pasting speech segments
Formant analysis- displays formant tracks of F1, F2 and F3
Pitch analysis
Filter tool- filters speech signal at cutoff frequencies specified by the user
Comparison tool- compares two waveforms using several spectral distance measures
Speech degradation- adds noise to the speech signal at an SNR specified by the us Platform: |
Size: 1180672 |
Author:wsw |
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Description: 本程序为一个用matlab实现的语音滤波程序，能计算并显示语音信号的幅频特性，巴特沃斯低通滤波器幅度和相位响应，引入噪声后的幅频特性，以及滤波后的DFT幅频特性。本程序附有语音文件以供试验。-This program is an implementation using matlab voice filtering procedures, can calculate and display the speech signal amplitude-frequency characteristics of Butterworth low-pass filter magnitude and phase response, after the introduction of noise, amplitude-frequency characteristics, as well as the filtered DFT amplitude-frequency characteristics. This procedure for testing with voice file. Platform: |
Size: 4096 |
Author:周峰 |
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Description: 提出一种有效的语音增强算法，该算法使用重叠相加FF-r滤波器进行谱幅度估计，给出噪声估计的两个均值参数，此参数适用于任何噪声．而且进行噪声估计无需检测语音间隙，无需VAD的运算过程，克服了单信道噪声
压缩算法的缺点．-An effective speech enhancement algorithm using the overlap-add FF-r filter in spectral amplitude estimation, given the noise mean estimate two parameters, this parameter applies to any noise. And no detection of speech to noise estimation space, without VAD computing process, overcome the single-channel noise suppression algorithm shortcomings. Platform: |
Size: 193536 |
Author:Decheng Yu |
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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 |
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Description: Used on cellular phones to filter out
background noise. Measure the signal to noise ratio of an audio input in realtime. Pass speech but give silence where the input is noise (hiss, snap crackle and pop).
Includes a tutorial on how to use statistics (correlation coefficient) to calculate the signal to noise ratio.
Realtime implementation using Texas Instruments TMS320C5510 DSK and TMS320C6713 DSK. Code mostly in C, except where it would be too slow and assembly language is used. Platform: |
Size: 1053696 |
Author:ivan |
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Description: 读入语音信号，加入噪声后将其滤除再还原声音-Adding a pulse noise to an existed speech then filt it with a specified FIR filter. Platform: |
Size: 314368 |
Author:刘进 |
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Description: • Plot the noisy speech signal both in time-domain and frequency-domain.
• 进行含噪语音信号的时频分析
• Design a proper filter to remove noise from the speech signal.
• 设计合适滤波器进行去噪
• Plot the clean signal both in time-domain and frequency-domain.
• 进行去噪后信号的时频分析
• Design a Reverberator using 4 Comb filters and 2 All-pass filters, as shown in the following Figure, to generate reverberated and reflected sound with the capability of passing it through an equalizer.
• 设计一个混响器（用四个梳状滤波器和两个全通滤波器（下图所示））来产生回声（通过一个均衡器（用于调整频率的补偿和衰减））
-• Plot the noisy speech signal both in time-domain and frequency-domain.
• 进行含噪语音信号的时频分析
• Design a proper filter to remove noise from the speech signal.
• 设计合适滤波器进行去噪
• Plot the clean signal both in time-domain and frequency-domain.
• 进行去噪后信号的时频分析
• Design a Reverberator using 4 Comb filters and 2 All-pass filters, as shown in the following Figure, to generate reverberated and reflected sound with the capability of passing it through an equalizer.
• 设计一个混响器（用四个梳状滤波器和两个全通滤波器（下图所示））来产生回声（通过一个均衡器（用于调整频率的补偿和衰减））
Platform: |
Size: 49152 |
Author:hanling |
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Description: 自适应噪声消除滤波器！是采用的语音信号！欢迎大家下载！-Adaptive noise cancellation filter! Is the use of the voice signal! Welcome to download!
Platform: |
Size: 1024 |
Author:luqin |
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Description:
采用winner的语音降噪实现代码，适用于语音和音频系统中，可直接调用。-Speech noise reduction using Winner filter implementation code, for voice and audio systems, can be called directly. Platform: |
Size: 1024 |
Author:梁小涛 |
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Search - speech noise filter