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I am designing an audio equalizer in Matlab, and I want to do it with a multirate setup, but I do not want to sample the input wave more than once, so if I have a sampling of say 50khz, I was wondering what the best way to decimate the signal would be in order to make it work. I plan on implementing this later to actual DSP chips, so thats is another reason I only want 1 input (ie 1 analog to digital converter). The bands are as follows:
I am using the remez exchange algorithm in order to achieve the various filters, the 1st and 7th filters will be a low and high pass, while 2 -6 will be bandpass filters. Thanks for any suggestions! Craig Roush
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Multi-rate Fir Filter Design
Started by ●November 4, 2002
Reply by ●November 7, 20022002-11-07
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You can do resampling in the digital domain also. So you wont need another A/D converter. Efficient implementation usually is done by combining the filtering and the downsampling stages. Doing a websearch on keywords like 'FIR decimation filter', 'poly phase filter' may give some useful results. Navan --- c r <> wrote: > > I am designing an audio equalizer in Matlab, and I > want to do it with a multirate setup, but I do not > want to sample the input wave more than once, so if > I have a sampling of say 50khz, I was wondering what > the best way to decimate the signal would be in > order to make it work. I plan on implementing this > later to actual DSP chips, so thats is another > reason I only want 1 input (ie 1 analog to digital > converter). The bands are as follows: > > 20-80hz > 80- 160Hz > 160-320 Hz > 320 - 800 hz > 800Hz - 2Khz > 2 - 10 Khz > 10 - 20khz > > I am using the remez exchange algorithm in order to > achieve the various filters, the 1st and 7th filters > will be a low and high pass, while 2 -6 will be > bandpass filters. > > Thanks for any suggestions! > > Craig Roush __________________________________________________ |
