fft

How to write fftshift and ifftshift in R? [closed]

由 爷，独闯天下 提交于 2019-12-19 03:50:33
问题 Closed . This question needs to be more focused. It is not currently accepting answers. Want to improve this question? Update the question so it focuses on one problem only by editing this post. Closed 3 years ago . In numpy , we have the following functions : import numpy from numpy.fft import fft2, ifft2, fftshift, ifftshift I would like to rewrite these functions in R. fft in R works just as fft or fft2 in python. Also for ifft2 , we have to do fft(,inverse=T) Now I would like to know how

How to calculate FFT using NAudio in realtime (ASIO out)

由 折月煮酒 提交于 2019-12-19 03:31:08
问题 I am programming clone of guitar (violin) Hero as a final project for this school year. The idea is to take input from my electric violin, analyse it via FFT, do some logic and drawing and output it through speakers. Perhaps some steps in parallel threads. I already have Asio low latency input-output implemented but I am having a great problem implementing realtime FFT. This is a code that sets up asioOut along with sampleAggregator. Sample aggregator should store samples that are added each

C# VS2019 调用FFTW FFT计算

由 妖精的绣舞 提交于 2019-12-19 02:57:01
申请空间时，请使用GCHandle.Alloc，这里只是调用DLL，并没有将其静态连接的 .lib 文件 将libfftw3-3.dll 文件放到项目中 //申请空间 //IntPtr In_intPtr= FFT.malloc(N_FFT*8); //IntPtr Out_intPtr= FFT.malloc(N_FFT * 8*2); GCHandle In_intPtr = GCHandle.Alloc(DataIn_y1, GCHandleType.Pinned); GCHandle Out_intPtr = GCHandle.Alloc(fftw_Complex, GCHandleType.Pinned); //正变换 IntPtr plan = FFT.plan_r2c_ld(N_FFT, In_intPtr.AddrOfPinnedObject(), Out_intPtr.AddrOfPinnedObject(), 0); //执行变换 FFT.execute(plan);//可能重复执行， FFT.execute(plan); //释放输入输出数组 FFT.destroy_plan(plan); //释放空间 In_intPtr.Free(); Out_intPtr.Free(); TTF public class FFT { #region FFTW 库调用申明

Displaying a double (frequency) that is updated constantly while recording with Android

由 若如初见. 提交于 2019-12-18 18:32:27
问题 I am building an android application that displays the Frequency of a sustained note with the FFT algorithm. I am using Jtransform methods. My issue currently is that I can't display the frequency on screen. The following code is the fft freqency calculation and the AsynchTask which should display the frequency in a text box import edu.emory.mathcs.jtransforms.fft.DoubleFFT_1D; public class Tuning extends Activity implements OnClickListener{ int audioSource = MediaRecorder.AudioSource.MIC; //

C# library to do fft and ifft? [closed]

由 允我心安 提交于 2019-12-18 17:56:15
问题 Closed. This question is off-topic. It is not currently accepting answers. Want to improve this question? Update the question so it's on-topic for Stack Overflow. Closed last year . is there a free c# library to do the fast fourier transform and its inverse? 回答1: I believe this is what you're looking for. 回答2: Both Math.NET and Aforce are based on the Exocortex.DSP library. http://www.exocortex.org/dsp 来源： https://stackoverflow.com/questions/1676294/c-sharp-library-to-do-fft-and-ifft

How to convert 16-bit PCM audio byte-array to double or float array?

由 ≯℡__Kan透↙ 提交于 2019-12-18 15:53:42
问题 I'm trying to perform Fast Fourier Transform on a .3gpp audio file. The file contains a small 5 second recording in 44100kHz from the phones microphone. Every Java FFT algorithm I can find only takes double[], float[] or Complex[] inputs, for obvious reasons, but I'm reading in the audio file in a byte-array, so I'm kind of confused as to where I go from here. The only thing I could find is the answer to a previous question: Android audio FFT to retrieve specific frequency magnitude using

Plotting FFT on octave

由 风格不统一 提交于 2019-12-18 15:00:27
问题 I know that FFT changes a function in the time domain to one showed in the frequency domain. However, when I try plotting said graph in the frequency domain, I can only get it to work properly by using the time as X-axis, when it was obviously supposed to be not that, but the frequency. Also, I can only get the amplitudes to match the ones in the original signal by dividing the y-axis by a certain integer. Why is that? Here's my code t=0:0.001:2 x=2*sin(20*pi*t) + sin(100*pi*t) subplot(2,1,1)

Fourier space filtering

由 让人想犯罪 __ 提交于 2019-12-18 13:34:28
问题 I have a real vector time series x of length T and a filter h of length t << T. h is a filter in fourier space, real and symmetric. It is approximately 1/f. I would like to filter x with h to get y. Suppose t == T and FFT's of length T could fit into memory (neither of which are true). To get my filtered x in python, I would do: import numpy as np from scipy.signal import fft, ifft y = np.real( np.ifft( np.fft(x) * h ) ) ) Since the conditions don't hold, I tried the following hack: Select a

How many FFTs per second can I do on my smartphone? (for performing voice recognition)

由 时光总嘲笑我的痴心妄想 提交于 2019-12-18 12:29:22
问题 I'm exploring voice recognition and DSP, and so I would like to implement a simple sound frequency analyzer on my smartphone (I have both an iPhone and a Samsung Nexus S running Android). I have done basic DSP in Matlab previously. From my understanding, I need to perform an FFT to get the fundamental frequencies of a signal. So now, I would like to sample the microphone at 44100 Hz. If I use a sliding window of sample size 512 with 50% overlap, that means I need to do an FFT every 256

MATLAB - Missing fundamental from an FFT [closed]

由 走远了吗. 提交于 2019-12-18 12:26:43
问题 Closed . This question needs to be more focused. It is not currently accepting answers. Want to improve this question? Update the question so it focuses on one problem only by editing this post. Closed 5 years ago . I am currently working on my fourth year project (computer science) which involves the automatic transcription of music -> sheet music. I am doing it in Matlab at the moment but will have to be converted to java at some stage. My problem: I have my program returning the correct
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