frequency-analysis

Converting Real and Imaginary FFT output to Frequency and Amplitude

人盡茶涼 提交于 2019-12-04 14:28:20
问题 I'm designing a real time Audio Analyser to be embedded on a FPGA chip. The finished system will read in a live audio stream and output frequency and amplitude pairs for the X most prevalent frequencies. I've managed to implement the FFT so far, but it's current output is just the real and imaginary parts for each window, and what I want to know is, how do I convert this into the frequency and amplitude pairs? I've been doing some reading on the FFT, and I see how they can be turned into a

FFT pitch detection for guitar string

拈花ヽ惹草 提交于 2019-12-04 12:33:32
I have simply pitch detection. Input (microphone) data are passed to fft routine, then I'm looking for a pitch with maximum value It means: Max(pow(data[i].getRe(), 2) + pow(data[i].getIm(), 2)) for 0<= i < SAmplesSize I need it for detection of guitar string's primary frequency. It works well for freq 440 hz (and maybe higher, i didn't check that) downto 250 hz. Below this value detected frequency is twice as high as it should be, ie. for 195 hz detected frequency is about 380 hz. It looks like it detects higher harmonics below 250 hz. For pure 195 hz tone it detects perfectly, but for quitar

How to get the fundamental frequency from FFT?

若如初见. 提交于 2019-12-04 04:59:24
问题 I am developing a flash guitar, and the only apparent method to discover the frequency of the mic's data looks to be using FFT. Nevertheless, after something like 30 hours of research I could not discover the best way to do that. Should I use Harmonic Product Spectrum (HPS), Cepstrum or Maximum Likelihood? The tuner will have to show frequencys from 25 to 3000Hz. Also, if someone knows about some open source code ready to find the fundamental frequency, please tell me. I can translate it to

How can I transfer a discrete set of data into the frequency domain and back (preferrably losslessly)

浪子不回头ぞ 提交于 2019-12-03 21:32:40
I would like to take an array of bytes of roughly size 70-80k and transform them from the time domain to the frequency domain (probably using a DFT). I have been following wiki and gotten this code so far. for (int k = 0; k < windows.length; k++) { double imag = 0.0; double real = 0.0; for (int n = 0; n < data.length; n++) { double val = (data[n]) * Math.exp(-2.0 * Math.PI * n * k / data.length) / 128; imag += Math.cos(val); real += Math.sin(val); } windows[k] = Math.sqrt(imag * imag + real * real); } and as far as I know, that finds the magnitude of each frequency window/bin. I then go

Matlab: Finding dominant frequencies in a frame of audio data

北战南征 提交于 2019-12-03 21:11:37
I am pretty new to Matlab and I am trying to write a simple frequency based speech detection algorithm. The end goal is to run the script on a wav file, and have it output start/end times for each speech segment. If use the code: fr = 128; [ audio, fs, nbits ] = wavread(audioPath); spectrogram(audio,fr,120,fr,fs,'yaxis') I get a useful frequency intensity vs. time graph like this: By looking at it, it is very easy to see when speech occurs. I could write an algorithm to automate the detection process by looking at each x-axis frame, figuring out which frequencies are dominant (have the highest

how can the noise be removed from a recorded sound,using fft in MATLAB?

老子叫甜甜 提交于 2019-12-03 06:06:52
问题 I want to remove noises from a recorded sound and make the fft of it finding fundamental frequencies of that sound, but I don't know how to remove those noises. I'm recording the sound of falling objects from different heights. I want to find the relation between the height and the maximum frequency of the recorded sound. [y,fs]=wavread('100cmfreefall.wav'); ch1=y(:,1); time=(1/44100)*length(ch1); t=linspace(0,time,length(ch1)); L=length(ch1); NFFT = 2^nextpow2(L); % Next power of 2 from

Awk: Words frequency from one text file, how to ouput into myFile.txt?

左心房为你撑大大i 提交于 2019-12-02 06:35:48
Given a .txt files with space separated words such as: But where is Esope the holly Bastard But where is And the Awk function : cat /pathway/to/your/file.txt | tr ' ' '\n' | sort | uniq -c | awk '{print $2"@"$1}' I get the following output in my console : 1 Bastard 1 Esope 1 holly 1 the 2 But 2 is 2 where How to get into printed into myFile.txt ? I actually have 300.000 lines and near 2 millions words. Better to output the result into a file. EDIT: Used answer (by @Sudo_O): $ awk '{a[$1]++}END{for(k in a)print a[k],k}' RS=" |\n" myfile.txt | sort > myfileout.txt Your pipeline isn't very

How to get the fundamental frequency from FFT?

一世执手 提交于 2019-12-02 06:24:09
I am developing a flash guitar, and the only apparent method to discover the frequency of the mic's data looks to be using FFT. Nevertheless, after something like 30 hours of research I could not discover the best way to do that. Should I use Harmonic Product Spectrum (HPS), Cepstrum or Maximum Likelihood? The tuner will have to show frequencys from 25 to 3000Hz. Also, if someone knows about some open source code ready to find the fundamental frequency, please tell me. I can translate it to ActionScript. Thanks! shams The simplest technique I've had success with is to use FFT and then perform

Improving frequency resolution of FFT output by limiting frequency range?

我怕爱的太早我们不能终老 提交于 2019-11-30 20:06:53
I am new to FFTs and signal processing, so hopefully this question makes sense and/or isn't stupid. I would like to perform spectrum analysis on a live audio signal. My goal is to find a good tradeoff between responsiveness and frequency resolution, such that I can take a guess at the pitch of the incoming audio in near-realtime. From what I've gathered about the math behind the Fourier transform, there is an inherent balance between sample size and frequency resolution. The bigger the sample, the better resolution. Since I am trying to minimize sample size (to attain the near-realtime

Improving frequency resolution of FFT output by limiting frequency range?

谁说我不能喝 提交于 2019-11-30 04:37:34
问题 I am new to FFTs and signal processing, so hopefully this question makes sense and/or isn't stupid. I would like to perform spectrum analysis on a live audio signal. My goal is to find a good tradeoff between responsiveness and frequency resolution, such that I can take a guess at the pitch of the incoming audio in near-realtime. From what I've gathered about the math behind the Fourier transform, there is an inherent balance between sample size and frequency resolution. The bigger the sample