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I don't have full info on how you are programming this, but I am
fairly sure that your problem is caused by unavoidable delay in the ADC, DSP and DAC components. The ADC takes some time to convert a signal. This delay may be unavoidable or may be in part due to the way the software works with the part. If you are using a standard ADC which requires a trigger to start conversion, then your software is triggering a conversion. The DSP reads the converted data and triggers a new conversion. If this process is repeated at the 20 KHz rate, the data is always one sample period old when read by the DSP. If you are using a sigma delta converter, then there is a fixed delay caused by the filters in the ADC. This is often several sample periods long. The DAC will have similar delays depending on the type. A standard converter will have about one sample period delay before the output is stable. A sigma delta converter will have a longer delay due to the filtering. Finally, depending on how you write your code, you may have very long delays in the DSP. Many applications use multiple buffers which are swapped as they are filled and emptied. This will create delays as long as your buffer size. Even if you don't have a buffer, you will likely to see a one sample delay depending on how your code transfers the data from the ADC to the DAC. So once you realize that it is hard to remove the fixed delay from such an application, you need to find out how long your delay is. You should be able to measure the fixed delay at several frequencies to verify that it is indeed a fixed delay. As long as it is fixed, this should be no problem. This will not distort pulses or any other waveshape. Most designers have a problem if the delay is anything other than a fixed delay. This is also called a linear phase delay implying that the delay varies linearly with frequency which is what you seem to be seeing. The best way to minimize this delay is to eliminate any of the sources of delay that you can. Then increase the sample rate to minimize the effect of any delays that you can't eliminate. If you really need a fixed phase delay in this process, then you will need to design a filter for the DSP that will compensate for the delay by making it longer, but a fixed phase. This will be not so easy and gets into very complex filter design. At 09:45 AM 4/21/00 , you wrote: >Hi all.. >I just started using the C31 not long ago; so i am still in the >learning process. There is something that I just cant seem to >understand! I have this program that just echos the input out. >However, when i change the frequency of the input, the phase changes >between the two signals (input and output). It keeps shifting as I >change frequency, although the two signals are still the same! Now.. >i thought that would be due to the delay in Calculations but from >what i understand that it is supposed to be very fast and I am >processing a signal that is less than 1 KHz with a sampling freqeuncy >of 20KHz! If you know why that is happening, please let me know and >how can I make the two signals have a fixed phase shift as I change >the frequency? Thank youuuu :) Rick Collins Arius - A Signal Processing Solutions Company Specializing in DSP and FPGA design Arius http://www.arius.com 4 King Ave 301-682-7772 Voice Frederick, MD 21701-3110 301-682-7666 FAX |