Search - frequency domain filter

Search - frequency domain filter - List

[Speech/Voice recognition/combine] math2 DL : 0: 一些数学处理函数，包括：用频域最小误差平方方法设计线性相位FIR低通数字滤波器、最小均方(LMS)算法的自适应数字滤波、离散小波变化函数-Deal with some mathematical functions, including: use of frequency domain methods of design of the smallest square error linear phase FIR low-pass digital filter, least mean square (LMS) algorithm for adaptive digital filtering, discrete wavelet function changes
Date : 2025-12-13 Size : 3kb User : chenjie
[Speech/Voice recognition/combine] FLMS.ZIP DL : 0: 这是一个很好的频域分块自适应滤波的程序，应用于回声消除上，并与NLMS自适应滤波做了运行速度的比较,FLMS比NLMS快几十倍。-This is a very good frequency-domain block adaptive filtering procedure applied to echo cancellation, and comparison with the NLMS adaptive filter for speed, FLMS several times faster than the NLMS.
Date : 2025-12-13 Size : 646kb User : 宋知用
[Speech/Voice recognition/combine] analoge_high_pass DL : 0: In oreder to compare Impulse Invariance method charactristics for transformation analogue filters to digital ones in highpass and bandpass and lowpass formats,here are 6 M-files that simulate a third order butterworth filter analyizing impulse response(time domain) and frequency analysis of analogue and transformed digital filters-In oreder to compare Impulse Invariance method charactristics for transformation analogue filters to digital ones in highpass and bandpass and lowpass formats,here are 6 M-files that simulate a third order butterworth filter analyizing impulse response(time domain) and frequency analysis of analogue and transformed digital filters
Date : 2025-12-13 Size : 1kb User : salar
[Speech/Voice recognition/combine] SimulinkFSK DL : 0: 题目要求： 1.录制一段自己的语音信号，并对录制的信号进行采样； 2.画出采样后的语音信号的时域波形和频谱图； 3.给定滤波器的性能指标，采用窗函数法和双线性变换法设计滤波器，并划出滤波器的频域响应； 4.用该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； 5.回放语音信号； 6.设计一个信号处理系统界面。-Subject to: 1. Recorded his voice for some signal, and recording the signal is sampled 2. Draw the sampled speech signal time-domain waveform and frequency spectrum 3. For a given filter performance, using window function bilinear transformation design method and filter, and set aside the filter frequency response 4. with the filter on the collection of signal filtering, to draw the filtered signal in time domain waveforms and frequency spectrum, and before and after filtering The signal contrast, analysis of signal changes 5. playback of audio signals 6. to design a signal processing system interface.
Date : 2025-12-13 Size : 2kb User : pengtianwei
[Speech/Voice recognition/combine] BasedonMATLABspeechsignalspectrumanalysisandfilter DL : 0: 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
Date : 2025-12-13 Size : 12kb User : 姚湘陵
[Speech/Voice recognition/combine] iirsinglefnoise DL : 0: 首先提取语音信号，并对其加单频噪声，设计IIR滤波器，滤除单频噪声，显示滤除前后时域波形的对比，并播放滤除后的语音信号。-First, extract the speech signal, and add single-frequency noise, design IIR filters to filter out single-frequency noise. Display the time-domain waveform before and after filter, and play the voice signal after filtering.
Date : 2025-12-13 Size : 1kb User : bryanchung
[Speech/Voice recognition/combine] yuyingfenxi DL : 0: 录制一段自己的语音信号，时间为10s左右，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；采用窗函数法或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语音信号进行滤波，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.-Record a voice signal their time was 10s or so, and the recorded signal is sampled draw sampled speech signal time-domain waveform and frequency spectrum using window functions or bilinear transformation design filters, and draw filter frequency response then use the filter of their own design collection of the voice signal filtering, to draw the filtered signal in time domain waveform and frequency spectrum, and comparing the signal before and after filtering, analysis of signal changes playback of audio signals .
Date : 2025-12-13 Size : 1kb User : wgh
[Speech/Voice recognition/combine] MATLAB-SINGAL-PROCESSING DL : 0: 对采集的语音信号进行滤波，谱分析，并进行前后时域和频域波形比较。可选择IIR和FIR滤波器进行高通，低通和带通滤波。-To obtain the voice signal filtering, spectrum analysis, and the time domain and frequency domain waveform and compared. Can choose IIR and FIR filter for qualcomm, low pass and bandpass filters.
Date : 2025-12-13 Size : 7kb User : shaw
[Speech/Voice recognition/combine] The-FIR-filter-design DL : 0: （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
Date : 2025-12-13 Size : 270kb User : smilence
[Speech/Voice recognition/combine] matlab DL : 0: 1. 给一段原始的语音信号（可以是自己录制的一段语音），加上一频率为3.8kHz的高频余弦噪声和频率为3.6kHz的高频正弦噪声（幅度自己可以选择），用窗函数设计一滤波器（要求最小阻带衰减为50dB）对加噪后的语音信号进行滤波，画出滤波器的频率响应曲线，画出滤波前后的时域图和频谱图。需要用到的函数： fir1 用窗函数设计FIR滤波器的函数 2. 用GUI设计一界面（如图1所示）完成如下功能： 1）输入一语音信号，画出语音信号的时域图和频谱图； 2）对语音信号加噪处理，画出加噪后的时域图和频谱图； 3）给定滤波器的性能指标，采用几种不同的方法设计滤波器对加噪信后进行滤波，画出各滤波器的频率响应曲线，画出滤波后的时域图和频谱图。 -1. to some of the original speech signal (which can be a voice you record yourself), plus a cosine frequency of 3.8kHz frequency noise and frequency of 3.6kHz frequency sinusoidal noise (in the range that they can choose), with a window function to design a filter (requires a minimum stop band attenuation is 50dB) after adding noise to the speech signal is filtered, the filter frequency response curve to draw, draw diagrams and time-domain spectrum before and after filtering. Need to use the function: function fir1 with window function design FIR filter design 2. Using a GUI interface (Figure 1) complete the following functions: 1) an input speech signal, the speech signal in the time domain to draw graphs and spectrum 2) adding noise to the speech signal processing, draw diagrams and time-domain plus noise spectrum after 3) to set the filter performance, using several different methods designed to filter letter after adding noise filtering, draw the frequency response curve of each
Date : 2025-12-13 Size : 1kb User : longteng
[Speech/Voice recognition/combine] chenxu DL : 0: （1）录制一段语音信号，完成对信号的采样，画出信号的时域波形和频谱图，确定信号的频谱范围；（2）给信号叠加噪声（噪声类型分为如下几种：a白噪声；b单频噪色（正弦干扰）；c多频噪声（多正弦干扰）；d其它干扰。），画出受噪声干扰的信号时域波形和频谱图；（3）采用窗函数法设计FIR低通滤波器，画出滤波器的频响特性图；（4）用所设计的滤波器对受噪声影响的信号进行滤波，画出滤波后语音信号的时域波形图和频谱图；（5）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，并与原始语音信号对比。((1) record a speech signal, complete the sampling of the signal, draw the time domain waveform and frequency spectrum of the signal, and determine the spectrum range of the signal; (2) signal superposition noise (noise type is divided into the following: a white noise, B single frequency noise (sinusoidal interference); C multifrequency noise (multi sinusoidal interference); d other interference) Noise signal, time domain waveform and spectrogram are drawn; (3) the FIR low-pass filter is designed by using the window function method, and the frequency response characteristic of the filter is drawn; (4) the designed filter is used to filter the signal affected by the noise, and the time domain waveform and spectrogram of the filtered speech signal are drawn; (5) compare the signals before and after filtering, analyze the change of the signal, replay the speech signal, and compare with the original speech signal)
Date : 2025-12-13 Size : 13kb User : 华