fft

问题 I am a no math guru here, so I want to ask anyone familiar with Digital Signal Processing, what is the best way of detecting real time peaks. I get about 30 frames/values a second and I've tried to implement the slope algorithm for detecting peaks, it worked OK, about 80% of the cases, but its really not good enough :(. From what I've searched one should use the Fast Fourier Transform, but I have no idea how to get started with it, perhaps I'm missing the general idea of how I should use FFT

问题 i have acquired a 5 minutes raw eeg from NEXUS 10 mark 2 equipment and it is giving me output in the matlab as 1 x 76800 row vector. as i understand , the sampling frequency choosen is 256 hz , hence it is giving me total 76800 sampling points. no wat i m perfoming N point FFT on this raw eeg signal. since N can only be power of 2 i am EXTRACTING 65536(2 ^16) SAMPLING POINTS FROM RAW EGG i.e. from 76800 points i have taken 65536. now i am not able to perform fft on the this vector (65536

问题 Here is a sample of the relevant code im working on: AudioRecord recorder = setupAudio(); recorder.startRecording(); SetupAudio method: public AudioRecord setupAudio() { AudioRecord recorder; minBufferSizeInBytes = AudioRecord.getMinBufferSize( RECORDER_SAMPLERATE, AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT); Log.e("MoverAudio","BufferSize: " + minBufferSizeInBytes); recorder = new AudioRecord(MediaRecorder.AudioSource.CAMCORDER, RECORDER_SAMPLERATE, AudioFormat.CHANNEL_IN

问题 So I've been working with FFT, and I'm currently trying to get a sound waveform from a file with FFT, (modify it eventually), but then output that modified waveform back to a file. I've gotten the FFT of the soundwave and then used an inverse FFT function on it, but the output file doesn't sound right at all. I haven't done any filtering on the waveform - I'm just testing out getting the frequency data and then putting it back into a file - it should sound the same, but it sounds wildly

问题 So I've been working with FFT, and I'm currently trying to get a sound waveform from a file with FFT, (modify it eventually), but then output that modified waveform back to a file. I've gotten the FFT of the soundwave and then used an inverse FFT function on it, but the output file doesn't sound right at all. I haven't done any filtering on the waveform - I'm just testing out getting the frequency data and then putting it back into a file - it should sound the same, but it sounds wildly

问题 I want to take two sounds that contain a dominant frequency and say 'this one is higher than this one'. I could do FFT, find the frequency with the greatest amplitude of each and compare them. I'm wondering if, as I have a specific task, there may be a simpler algorithm. The sounds are quite dirty with many frequencies, but contain a clear dominant pitch. They aren't perfectly produced sine waves. 回答1: Given that the sounds are quite dirty, I would suggest starting to develop the algorithm

问题 I am trying to do a quick spectral analysis on the streaming audio data to capture vowels (something like JLip-sync). Using PyAudio to capture the voice data in small chunks (1024) for short durations (0.0625 sec.). Using numpy.fft for the analysis, and to get rid of leakage using numpy.hanning window. I am using 4096*4 as the sampling rate (not 44100 or 22050, and open to discussion as well; 4096*4 being nearest to 22050). Considering the frequencies I am interested in (ranging from 300 Hz

问题 Here is the scenario: using a spectrum analyzer i have the input values and the output values. the number of samples is 32000 and the sampling rate is 2000 samples/sec, and the input is a sine wave of 50 hz , the input is current and the output is pressure in psi. How do i calculate the frequency response from this data using MATLAB, using the FFT function in MATLAB. i was able to generate a sine wave, that gives out the the magnitude and phase angles, here is the code that i used: %FFT

问题 I have an array of audio data I am passing to a reader: recorder.read(audioData,0,bufferSize); The instantiation is as follows: AudioRecord recorder; short[] audioData; int bufferSize; int samplerate = 8000; //get the buffer size to use with this audio record bufferSize = AudioRecord.getMinBufferSize(samplerate, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT)*3; //instantiate the AudioRecorder recorder = new AudioRecord(AudioSource.MIC,samplerate, AudioFormat.CHANNEL

问题 I have to use FFT to analyse the frequency of an audio file. But I don't know what the input and output is. Do I have to use 1-dimension, 2-dimension or 3-dimension array if I want to draw the spectrum's audio file? And can someone suggest me library for FFT on J2ME? 回答1: @thongcaoloi, The simple answer regarding the dimensionality of your input data is: you need 1D data. Now I'll explain what that means. Because you want to analyze audio data, your input to the discrete Fourier transform