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?
Giving large no of samples to KissFFT
431 Views Asked by Bhavin Chowksi At
1
There are 1 best solutions below
Related Questions in AUDIO
- Play multiple audio files in a slider
- Unity3d AudioSource not creatable
- JavaFX can't play mp3 files
- iPhone simultaneous sound output
- Phonegap Build App - Play Audio
- HTML5 Audio pause not working
- Java boolean play button issue (play over and over again with each click)
- import a sound externally or from the library? AS3
- Set audio source
- Saving a sound bite as a ringtone
- Using OnAudioFilterRead with playOnAwake
- Audio recorded with Samsung does not play on iOS
- fftw of 16bit Audio :: peak appearing wrong at 2f
- How to Export an audio file with effect in iOS
- Tried multiple solutions onsite, none worked: Play <audio> on Konami code
Related Questions in SIGNAL-PROCESSING
- Calculate energy for each frequency band around frequency F of interest in Python
- convert sound to list of phonemes in python
- Why is there a difference in magnitude response between scipy.filtfilt and scipy.lfilter?
- Image 2x downsampling with Lanczos filter
- Simple Python Median Filter for time series
- FFT Fundamental frequency calculation from LomontFFT
- Daubechies orthogonal wavelet in python
- fftw slight peak inaccuracy/drifting
- Zoom in on np.fft2 result
- How can I find process noise and measurement noise in a Kalman filter if I have a set of RSSI readings?
- Giving large no of samples to KissFFT
- FFT coefficients using python
- SignalGenerator class at naudio library - duration time play
- Extract binary data stream from audio signal
- FFT: find and cut noisy 50Hz in signal
Related Questions in FFT
- 3D FFT with data larger than cache
- FFT Filtering of signal
- Changing the amount of points changes the result of the fft
- Finding peak frequency in a complex signal using Matlab
- fftw of 16bit Audio :: peak appearing wrong at 2f
- Detect repetition in text string / copied text
- FFT in Arrayfire is slower than in MATLAB
- FFT Fundamental frequency calculation from LomontFFT
- Polynomial multiplication in M2(R)?
- fftw slight peak inaccuracy/drifting
- Zoom in on np.fft2 result
- Giving large no of samples to KissFFT
- cross correlation using fft producing inaccurate results
- Weird but close fft and ifft of image in c++
- How to get complex64 output from numpy.fft?
Related Questions in KISSFFT
- Giving large no of samples to KissFFT
- fftshift before calculating fourier transform: Matlab
- Why does my KISS FFT plot show duplicate peaks mirrored on the y-axis?
- KissFFT output of kiss_fftr
- KissFFT forward / inverse is outputting noise, why?
- KissFFT (kiss_fftr to kiss_fftri) - How to reconstruct the original signal?
- How to interpret the result from KissFFT's kiss_fftr (FFT for a real signal) function
- Using KISS_FFT on a microcontroller
- how to use kiss fft in Visual Studio 2010
- how can I define plan and executing it in Kiss FFT
- Create spectrum from KissFFT and QAudioProbe
- Why is the kiss_fft's forward and inverse radix-4 calculation different?
- kiss_fftr followed by kiss_fftri (with very large window size) does not return the input signal
- LNK2019: unresolved external symbol _kiss_fftr_alloc referenced in function "void __cdecl mainfunc(void)
- How to perform FFT2D (Fast Fourier Transform 2D) R, G, B color component
Trending Questions
- UIImageView Frame Doesn't Reflect Constraints
- Is it possible to use adb commands to click on a view by finding its ID?
- How to create a new web character symbol recognizable by html/javascript?
- Why isn't my CSS3 animation smooth in Google Chrome (but very smooth on other browsers)?
- Heap Gives Page Fault
- Connect ffmpeg to Visual Studio 2008
- Both Object- and ValueAnimator jumps when Duration is set above API LvL 24
- How to avoid default initialization of objects in std::vector?
- second argument of the command line arguments in a format other than char** argv or char* argv[]
- How to improve efficiency of algorithm which generates next lexicographic permutation?
- Navigating to the another actvity app getting crash in android
- How to read the particular message format in android and store in sqlite database?
- Resetting inventory status after order is cancelled
- Efficiently compute powers of X in SSE/AVX
- Insert into an external database using ajax and php : POST 500 (Internal Server Error)
Popular Questions
- How do I undo the most recent local commits in Git?
- How can I remove a specific item from an array in JavaScript?
- How do I delete a Git branch locally and remotely?
- Find all files containing a specific text (string) on Linux?
- How do I revert a Git repository to a previous commit?
- How do I create an HTML button that acts like a link?
- How do I check out a remote Git branch?
- How do I force "git pull" to overwrite local files?
- How do I list all files of a directory?
- How to check whether a string contains a substring in JavaScript?
- How do I redirect to another webpage?
- How can I iterate over rows in a Pandas DataFrame?
- How do I convert a String to an int in Java?
- Does Python have a string 'contains' substring method?
- How do I check if a string contains a specific word?
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 starting with a 50% overlap, so the first FFT would be for samples 0..4095, the second for 2048..6143, etc.