Hi, Multiply any FIR coefficient set with this sequence [...-1, +1, -1, +1...] and in effect you would mirror image the frequency response of...

Hi, Multiply any FIR coefficient set with this sequence [...-1, +1, -1, +1...] and in effect you would mirror image the frequency response of the filter around fs/4. In your case if [b0, b1] are the filter coefficients for a LPF then [b0, -b1] would translate it to a HPF. [...-1, +1, -1, +1...] is a cosine signal cos(pi*n) (Cosine wave of frequency fs/2 sampled at fs). Multiplying this with ...

Hi, I have a problem on frequency response. Suppose you have H(z)=B(z)/A(z), the transfer function of a digital filter. How do you calculate...

Hi, I have a problem on frequency response. Suppose you have H(z)=B(z)/A(z), the transfer function of a digital filter. How do you calculate the ANALYTIC expression of the filter's impulse response? Meaning, how do you get the analytic expressions for the amplitude response and the phase response? Fla.

Hi everyone, I have a question about filtering a noise signal in MATLAB. I have a corrupted sound file and I loaded it in MATLAB and plotted...

Hi everyone, I have a question about filtering a noise signal in MATLAB. I have a corrupted sound file and I loaded it in MATLAB and plotted the FFT of the signal. From the FFT, I found that there are two peaks present at + & - 44.04kHz at a magnitude of 2.029*10^5. Now, I need to design a filter to filter out the noise in the signal. I know that in MATLAB, I need to first define the tra...

Here's an article that may help. I admit that I only briefly looked at it, but the main emphasis of this thesis is "...dedicated to multiple...

Here's an article that may help. I admit that I only briefly looked at it, but the main emphasis of this thesis is "...dedicated to multiple fundamental frequency (F0) estimation". http://sp.cs.tut.fi/publications/theses/doctoral/Klapuri2004.pdf Aishwarya Venkataraman wrote: > > HI, > > We re working on a project dealing with south Indian > music signals. > We re right now stuck wi

I'm designing a parametric equalizer using a spartan 3e starter board and a TI PCM codec I wired to it, running at 48kHz. I decided to use the...

I'm designing a parametric equalizer using a spartan 3e starter board and a TI PCM codec I wired to it, running at 48kHz. I decided to use the biquad IIR filter core from opencores.org which accepts 5 16bit normalized coefficients. I'm using http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt as a guide for calculating coefficients. (this is not platform specific and applicable for dsp chips to...

Hello, I am looking for a way to obtain the lowest group delay. The specifications are; Lowpass filter fpass=1kHz with 3dB...

Hello, I am looking for a way to obtain the lowest group delay. The specifications are; Lowpass filter fpass=1kHz with 3dB Ripple fstop=4kHz Attenuation 60dB (between 4kHz and 20kHz or between 4kHz and 97.5kHz) with sampling frequency: 195kHz up to 1kHz linear phase and constant group delay (needed) If I use Generalized Equiripple FIR filter of Matlab fda toolbox, the minimum orde...

Hi all, I recenly join this group and i hope this will be a good plateform for me to share my ideas with others. I am doing SNR estimation with...

Hi all, I recenly join this group and i hope this will be a good plateform for me to share my ideas with others. I am doing SNR estimation with color noise for OFDM systems.i am confused about somthings so i am writing here.i am very thanksful for your kind replies. I take 256 bit symbol after adding cyclic prefix 'cp=1/4' its length becomes 320. For color noise estimation i divide the whole...

Hi, We are implementing an mpeg1 audio encoder in a fixed point dsp but we have problems in migrating the psychoacustic model to fixed point...

Hi, We are implementing an mpeg1 audio encoder in a fixed point dsp but we have problems in migrating the psychoacustic model to fixed point (all the log al pow instructions). Can anybody help us?. Regards, Juan

Hi all, I have designed a FIR filter which has 60 DB SNR.Also I designed IIR filter at 60 DB SNR. FIR taps is around 700 whereas IIR is 12. So...

Hi all, I have designed a FIR filter which has 60 DB SNR.Also I designed IIR filter at 60 DB SNR. FIR taps is around 700 whereas IIR is 12. So I want to design IIR.But I want linear phase. How can I achieve linear phase using IIR filters. Thank you, Abhijith

Hi! I'm new on dsp and i have some doubt. If i want to implemment some simple audio processor on a texas dsp(input 12 or maybe up to 16...

Hi! I'm new on dsp and i have some doubt. If i want to implemment some simple audio processor on a texas dsp(input 12 or maybe up to 16 bits, with a few effects and features like equalizer, chorus, delay, etc.), which dsp would be appropiate? For example, the 6000 floating point line is too much? It would be nice with the 2000 or 3000 type? I would like to have some estimation of the amoun...