fft

问题 I wanted to find the 4096 point DFT of an audio signal of duration 10 seconds with sampling rate 44100 Hz. Hence there are 441000 input samples. But KissFFT takes only up to 4096 as input size. How to go about finding FFT of such a large signal? 回答1: The power spectrum of most real-world audio signals (speech, music, etc) is time-varying, so typically you calculate a series of short-term FFTs using overlapping windows, to produce a sequence of power spectra, aka a spectrogram . I suggest

问题 I am currently involved in a project in which I want to find the average amplitude for given audio data in any given AAC file. I am currently reading the file as an array buffer and passing into an Uint8Array. var dataArray = new Uint8Array(buffer) Then I set up two arrays, one real(containing the audio data) and one imaginary(containing all zeros), and pass them into an FFT. The audio data is then placed into a new array such that the numbers within the array are no longer treated as

问题 I have a oscillator bank made in SuperCollider which receives phases and amplitudes from python via OSC. However the results don't sound correct at all. At first I thought the problem is in my SuperCollider code, but now I'm beginning to doubt my FFT function and normalization, here's my code: def readNormalize(length, location,sample): samplerate, data = wavfile.read(location) a = data.T[0] # first track of audio c = fft(a[sample:], length) ownSum = 0; length = int(length/2) for i in range(0

问题 I am trying to apply filter to image using 2D FFT. I have filter matrix (convolution kernel etc), but if I want to use fft I need to expand filter and picture matrix to power of 2 square matrix. First I expand it with zeros (so matrix of picture and filter would be in top left corner of big power of 2 matrix) and I got this result Then I tried what I did with picture matrix when I was applying convolution without fft, padded picture matrix with filter dimension/2 from each side, and than that

问题 Merged with Power Spectral Density from jTransforms DoubleFFT_1D. I'm using Jtransforms java library to perform analysis on a give dataset. An example of the data is as follows: 980,988,1160,1080,928,1068,1156,1152,1176,1264 I'm using the DoubleFFT_1D funtion in jTransforms. The data output is as follows: 10952, -152, 80.052, 379.936, -307.691, 12.734, -224.052, 427.607, -48.308, 81.472 I'm having trouble interpreting the output. I understand that the first element in the output array is the

问题 In this discussion, the result of fft is indices (0:N-1). fftshift simply converts that to [(N/2:N-1) (0:(N/2-1))]. I want to convert original range (O:N-1) to (t/N: t/N + 1), where t is time and assume integer and divisibel by N . I am using the Galois vectors as my datatype. Is this possible with built-in functions in Matlab? How can you achieve it in Matlab? 回答1: In general, given a data vector, if you want to shift the range from 0:N-1 to [a:N-1 0:a-1] for some a (0<=a<=N), you can do it

问题 I'm trying to apply a cuFFT, forward then inverse, to a 2D image. I need the real and complex parts as separate outputs so I can compute a phase and magnitude image. I haven't been able to recreate the input image, and also a non-zero phase is returned. In particular I am unsure if I'm correctly creating a full-size image from the reduced-size cuFFT complex output, which apparently stores only the left side of the spectrum. Here's my current code: // Load image cv::Mat_<float> img; img = cv:

问题 I work for an audio recorder app and would like to add gain control but I don't know how to do it. I have access to audio buffer but I don't know how to make the sound louder or slower. I see another apps that have a seekBar for selecting the decibels from a specific range. aRecorder.read(buffer, 0, buffer.length); fWriter.write(buffer); // Write buffer to file Please tell me what should I do to implement gain control in my app. Thanks. 回答1: Changing the volume is simple, just multiply all

问题 I'm trying to extract information from a sound file in order to use it in a video classification algorithm I'm working on. My problem is that I don't know how to work exactly with audio files in Matlab. Below is what I need to accomplish: open the audio file and get the sampling rate/frequency I need to work on a window of 2 seconds so I have to loop over the file and get each 2 seconds as a window and then do the ftt (Fast-Fourier-Transform) on each window. After that it is my turn to use

问题 Im trying to apply the fourier phase shift theorem to a complex signal in R. However, only the magnitude of my signal shifts as I expect it. I think it should be possible to apply this theorem to complex signals, so probably I make an error somewhere. My guess is that there is an error in the frequency axis I calculate. How do I correctly apply the fourier shift theorem to a complex signal (using R)? i = complex(0,0,1) t.in = (1+i)*matrix(c(1,0,0,0,0,0,0,0,0,0)) n.shift = 5 #the output of fft