Akash_DSP wrote:>> Akash_DSP wrote: >>>> Akash_DSP wrote: >>>>>> Akash_DSP wrote: >>>>>> >>>>>> ... >>>>>> >>>>>>> Hey Jerry, >>>>>>> >>>>>>> thanks for the reply. >>>>>>> >>>>>>> Yes, the periodic vairation is sinusoidal. This means that if I > give >>>>> an >>>>>>> audio input with a frequenct spectrum (<20 KHz), frequencies that >>> will >>>>> have >>>>>>> come out of the D/A will have different delays. >>>>>> Different delays for different frequencies, of different delays for >>> the >>>>>> same frequency at different times? >>>>>> >>>>>> ... >>>>>> >>>>>> Jerry >>>>>> -- >>>>>> Engineering is the art of making what you want from things you can >>> get. >>>>> Hey Jerry, >>>>> >>>>> it is different delays for different frequencies at the same time. >>>> It seems that you have discovered the dispersion of the anti-alias >>>> filter. That is normal. >>>> >>>> Jerry >>>> -- >>>> Engineering is the art of making what you want from things you can > get. >>> Hey Jerry, >>> >>> you mean to say that if I give an audio input and observe at the > output >>> that each frequency comes out with a different delay. That is normal? >>> >>> If you program your C6416 or c6713 do you see similar behaviour? >> In order to sample successfully, there must be no components of the >> sampled signal as high as half the sampling frequency. The use of an >> anti-alias low-pass filter in the signal path before the sampler is so >> nearly universal that many codecs have a filter built in. Being analog, >> the a-a filter will affect the phase. The designer is responsible for >> correcting any phase distortion that interferes with proper the function > >> of his design. >> >> Examining the data sheet should show you how much phase shift to expect. > >> Once you plot the filter's phase/frequency characteristic, you will know > >> if what you see is normal. >> >> Oversampling is an effective way to minimize phase shift in the band of >> actual interest. After sampling, the signal can be further filtered >> digitally, then decimated. >> >> All real design requires balancing competing objectives; that is an art. >> >> Jerry >> -- >> Engineering is the art of making what you want from things you can get. >> ??????????????????????????????????????????????????????????????????????? > > > Hey Jerry, > > thanks for the descriptive mail. I really appreciate it. > > I understand aliasing thats why in the first message I wrote that my > sampling frequeny is 44.1KHz and I am looking at frequency spectrum 0 > -20Khz (Following the Nyquist Criteria). I also understand the use of anti > aliasing filter. > > But I am seeing phase delay just moving +- 1ooHz . e.g. there is a big > phase difference b/w the input frequencies 16.0 KHz and 16.10 Khz. 16.0 Khz > and 17.7Khz will then have the same delay. As it is periodic delay. > > I am unable to find the reason for this as this is what we normally dont > see in a DSP chip.You're right: that isn't normal. The problem might be with the equipment used to measure the phase or the way you use it. I can more readily imagine that than I can something funny happening digitally. Do you have a two-channel oscilloscope available? If you do, sync on the codec's input and observe input and output simultaneously as you slowly vary the frequency. You can also use a single-channel scope. Apply the input as the horizontal signal and the output as the vertical. You will see a straight line when the signals are in phase and an ellipse when they're not. You can actually measure the phase angle with this setup, but I won't get into those details unless someone asks. Jerry -- Engineering is the art of making what you want from things you can get.
AIC23 _ High Pass Filter_BIG_FUNDAMENTAL_PROBLEM
Started by ●July 30, 2008
Reply by ●July 31, 20082008-07-31
