This question already has answers here: 开发者_如何学编程Closed 10 years ago. Possible Duplicate:
I\'m trying to track and analyze changes to a received signal due to changes in the channel using GNU Radio (also using Ettus USRPs). I would like to write a program using GNU Radio to be able to keep
I just want to clarify on the difference between certain \'implementat开发者_运维技巧ions of FFT\'. Ive read that there are 1D FFTs and then there are 2D FFTs and others.
All suggestions and links to relevant info welcome here. This is the scenario: Let us say I have a .wav file of someone speaking (and therefore all the samples associated with it).
I am trying to understand the FFT algorithm and so far I think that I understand the main concept behind it. However I am confused as to the difference between \'framesize\' and \'window\'.
Ive been experimenting with the FFT algorithm. I use NAudio along with a working code of the FFT algorithm from the internet. Based on my observations of the performance, the resulting pitch is inaccu
I have a stream of binary data and want to convert it to raw waveform sound data, which I can send to the speakers.
I have two audio recordings of a same signal by 2 different microphones (for example, in a WAV format), but one of them is recorded with delay, for example, several seconds.
I am developing a system as an aid to musicians performing transcription. The aim is to perform automatic music transcription (it does not have to be perfect, as the user will correct glitches / mista
Using FFTW I have been computing the FFT of normalized .wav file data. I am a bit confused as to how I should normalise the FFT output, however. I have been using the method which seemed obvious to me
加载中,请稍侯......