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Re: How to impoverish the FFT of a sound and keep the same sampling rate at once ?
inBernhard- > thanks for pointing this out. I agree in most of what you say. > > My intention, however, was to keep things as simple as...
Bernhard- > thanks for pointing this out. I agree in most of what you say. > > My intention, however, was to keep things as simple as possible. > > Besides, a broad variety of sigma-delta ADCs has the antialiasing filter > incorporated. In combination of a common analog front end including OP-amp > based decoupling stage and preamplifier which cannot handle frequencies much > higher th
Tip on developer kit with low-delay (<1ms) codec
inHi! I've been working with a 6713DSK but realized that the delay introduced by the codec is far to large (about 1.25 ms @ 22.05 ksps). I'm...
Hi! I've been working with a 6713DSK but realized that the delay introduced by the codec is far to large (about 1.25 ms @ 22.05 ksps). I'm trying to find a developer kit (doesn't care about manufacturer) that uses a codec/ADC-DAC which doesn't introduce more than, say, 0.5- 1 ms delay. I guess this implies a SAR-ADC instead of a Sigma-Delta approach. However, finding a complete develop...
How to impoverish the FFT of a sound and keep the same sampling rate at once ?
inHi everyone ! I'm a beginner in audio DSP applications. I'm a student working in the field of sound quality. In this framework, I'm facing...
Hi everyone ! I'm a beginner in audio DSP applications. I'm a student working in the field of sound quality. In this framework, I'm facing the following issue : I'd like to impoverish the frequency content of some 4-seconds recorded sounds (fs=44100hz, quality 16 bits), i.e. increase the frequency step Df of their FFT spectrum (for example 2Hz instead of 0.25Hz initially), while keepi...
Guitar Valve Preamp with DSP
inHi. I've working with al TMS320C613 DSK building some guitar effects. I was suscesfully whit all of these but I had no idea about build a valve...
Hi. I've working with al TMS320C613 DSK building some guitar effects. I was suscesfully whit all of these but I had no idea about build a valve guitar preamp on DSP. In other forums, people said I have to decimate, interpolate and check out curves in the valve 12AX7's datasheet and process the signal with that. The thing is .... I'm still have not much idea about it. Is someone could give...
Window for STFT analysis of low SNR signals
inHello, My question concerns the windowing in the STFT analysis. I deal with some specific signals, i.e. archival recording, which have poor SNR...
Hello, My question concerns the windowing in the STFT analysis. I deal with some specific signals, i.e. archival recording, which have poor SNR - around 20 or 30dB. Thus I wonder if there is any point of using windows with high side-lobe attenuation - higher then the signals? SNR? Thanks in advance for any clues Regards, PM
Generating Chirp/LFM by using IIR filters
inHelp needed in designing chirp generator and pulse compressor using iir filters (this amounts to using an IDT, I guess). I want to generate...
Help needed in designing chirp generator and pulse compressor using iir filters (this amounts to using an IDT, I guess). I want to generate chirp using an IIR filter. I want to send a pulse to the iir filter and expect an output which is a chirp signal starting at f0 and ending at f1 in time duration t0. Also, I want to use the iir filter to generate pulse when fed with the above chirp. ...
Linear Prediction Coding in Speech Synthesis
inHi, I am using Linear prediction coding technique for my speach synthesizer here is the process i am using. I pass my speech samples to...
Hi, I am using Linear prediction coding technique for my speach synthesizer here is the process i am using. I pass my speech samples to autocorrelation function to get autocorrelation. then this autocorrelation sequence is passed to levinson durbin recursion method to obtain filter parameters or filte coefficients. then these coefficents are used to obtain the original speech back i...
Re: Digest Number 442
Hi PrzeM, Yes, it is very possible to calculate the window-modified spectrum, which is a convolution of the data spectrum and the window...
Hi PrzeM, Yes, it is very possible to calculate the window-modified spectrum, which is a convolution of the data spectrum and the window spectrum. However, the temporal windows were invented to save efforts, instead of performing convolution in the freq domain perform multiplication in the time domain. Regards, Andrew > 2. Hann windowed FFT from 'rectangular' one ? > Posted b
Hann windowed FFT from 'rectangular' one ?
inHello, This is my first post in this group so let me say hi to everybody :) ... and of course I have a question, probably simple but...
Hello, This is my first post in this group so let me say hi to everybody :) ... and of course I have a question, probably simple but ... (how) can I get hann-windowed complex FFT values from the plain (unwindowed) complex FFT ? In other words, having complex spectra X = fft(x, N), can I get hann-windowed complex spectra Xh from X ? Thanks in advance, Best regards, PrzeM
SNR estimation or how to measure signal level
inHi all, I would really appreciate your ideas on something... Ok, I am building a speech recognition system which uses a mic array to...
Hi all, I would really appreciate your ideas on something... Ok, I am building a speech recognition system which uses a mic array to capture the sound. The signal coming in is not only speech but it's also affected by noise and room reverberation. I need to see what's the effect of reverberation alone so I mixed the speech signal from a close-talking microphone (so no noise and reverberati...
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