hi all,can some one plz tell me that how phase coherent summation works in polyphase filters? ?Here is some stuff i found in a paper but its not...

hi all,can some one plz tell me that how phase coherent summation works in polyphase filters? ?Here is some stuff i found in a paper but its not very clear..can anyone he;lp "The alias terms in each path of the M-path filter exhibit unique phase profiles due to their distinct center frequencies and the time offsets of the different down-sampled time series delivered to each port. These time of...

Hi, I've measured the impulse response of a loudspeaker, and considered as a FIR. Plotting its zeros, I see that nearly all the zeros are very...

Hi, I've measured the impulse response of a loudspeaker, and considered as a FIR. Plotting its zeros, I see that nearly all the zeros are very close to the unit circle...so that a little error in precision make the filter unstable. I wonder why it is so. Changing a little bit the measurement position doed not involve any unstability of the filter...nor the filtering with FIR is unstable... ...

Hi, I am a newcomer in the digital audio area & previously I worked in Video Decoders. I have embarked on my first project of...

Hi, I am a newcomer in the digital audio area & previously I worked in Video Decoders. I have embarked on my first project of WMA10(Windows media audio 10). I had the reference codes & specifications for WMA9 & WMA10. Basically I need to find the difference between these two versions & then start the coding of difference on processor which I had the assembly language co...

I'm implementing a DTMF detector using the Goertzel Algorithm. I have raw simulations built using Excel (I know...) and am getting some...

I'm implementing a DTMF detector using the Goertzel Algorithm. I have raw simulations built using Excel (I know...) and am getting some interesting results. I don't see a lot of differences in the frequency response using 10, 8 or 1 bit ADC's and floating point arithmetic. I would have expected to see significantly poorer performance with the 1-bit. I also ran some simulations using float a...

hiya, This question centers around the difference between having an odd or even number of taps in an FIR filter. I'm trying to determine...

hiya, This question centers around the difference between having an odd or even number of taps in an FIR filter. I'm trying to determine the coefficients of and FIR filter in mathematica. To do this i've used what i think is called the windowing method, where I take my ideal frequency response and transform it from frequency space into time space with a DTFT to get an impulse response. ...

Helloo, I am trying to understand the synthesis filter bank in the LIBMAD. I could not understand in what order the D coefficients are used...

Helloo, I am trying to understand the synthesis filter bank in the LIBMAD. I could not understand in what order the D coefficients are used to obtain the PCM samples. Also,half of these coefficients have some relation to the other half. Then, why all the coefficients are being used in D.dat file? What is the significance of even and odd pointers? How is the windowing operation between ...

Hi everybody, I am working on designing a digital hearing aid as my MSc Project and am facing severe difficulties in removing noise from speech....

Hi everybody, I am working on designing a digital hearing aid as my MSc Project and am facing severe difficulties in removing noise from speech. There are loads of stuff about VAD on the web but all of them explain only the theory. I am at a very cruicual stage because time is running out. Can anybody help me in this regard. Plz its urgent. Thanks in advance.

This question is not a real problem. It is just something which makes my head so confused and I really need a relief :) I tried to get a...

This question is not a real problem. It is just something which makes my head so confused and I really need a relief :) I tried to get a frequency response of an unknown system using its input-output-data. Input-data is PRBS-signal. It worked really good. My PRBS was length of 2000 and I used 1 period. I obtained the frequency response with FFT(output,one period)/FFT(input,one period). What ma...

I'm working on a DSP-based system for audio noise cancellation, and as a result I need to do the microphone-> ADC-> DSP-> DAC-> speaker loop...

I'm working on a DSP-based system for audio noise cancellation, and as a result I need to do the microphone-> ADC-> DSP-> DAC-> speaker loop quickly to output the anti-noise value before the noise input I sampled is gone. To date, I have used a SAR architecture ADC sampling at 20kHz. At the beginning of my 50us sample period, I tell the ADC to start its conversion. I get the digital value after 4

Hi, I'm new to DSP and just wrote a bunch of functions in C to process some audio data. I implemented a lowpass filter by convolving a...

Hi, I'm new to DSP and just wrote a bunch of functions in C to process some audio data. I implemented a lowpass filter by convolving a 256-sample audio sequence with a sinc function from -64 to 64 (129 samples) To convolve I do the fft of both functions and multiply their ffts, and then do the inverse fft of the product, and discard extra samples to have only 256 samples back. How...