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EE381K-14 Multidimensional DSP Multidimensional Resampling Lecture by Prof. Brian L. Evans Scribe: Serene Banerjee Dept. of Electrical and Comp. Eng. The University of Texas at Austin
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7 - 2 M x[n]x[n]xd[n]xd[n] One-Dimensional Downsampling Downsample by M –Input M samples with index –Output first sample (discard M–1 samples) Discards data May cause aliasing k i is called a coset k i ={0, 1, …, |M|-1}
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7 - 3 One-Dimensional Downsampling NN N 1 Xc(j)Xc(j) 1/T X ()X () T X d ( ) T’ Downsampling by M generates baseband plus M-1 copies of baseband per period of frequency domain Sample the analog bandlimited signal every T time units Aliasing occurs: avoid aliasing by pre- filtering with lowpass filter with gain of 1 and cutoff of /M to extract baseband Fig. 3.19(a)-(c) Oppenheim & Schafer, 1989. M=3
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7 - 4 One-Dimensional Upsampling Upsample by L –Input one sample –Output input sample followed by L–1 zeros Adds data May cause imaging L x[n]x[n]xu[n]xu[n]
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7 - 5 One-Dimensional Upsampling NN N 1 Xc(j)Xc(j) 1/T X ()X () T L 1/T X u ( ) = X(L ) T’ L L L L Upsampling by L gives L images of baseband per 2 period of Sample the analog bandlimited signal every T time units Apply lowpass interpolation filter with gain of L and cutoff of /L to extract baseband L 1/T = L/T X i ( ) L T’ Fig. 3.22 Oppenheim & Schafer, 1989.
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7 - 6 1-D Rational Rate Change Change sampling rate by rational factor L M -1 –Upsample by L –Downsample by M Aliasing and imaging Change sampling rate by rational factor L M -1 –Interpolate by L –Decimate by M Interpolate by L –Upsample by L –Lowpass filter with a cutoff of /L (anti-imaging filter) Decimate by M –Lowpass filter with a cutoff of /M (anti-aliasing filter) –Downsample by M
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7 - 7 1-D Resampling of Speech Convert 48 kHz speech to 8 kHz –48 kHz sampling: 24 kHz analog bandwidth –8 kHz sampling: 4 kHz analog bandwidth –Lowpass filter with anti-aliasing filter with cutoff at /6 and downsample by 6 Convert 8 kHz speech to 48 kHz –Interpolate by 6
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7 - 8 160 LPF 0 = /160 147 160 f s 160 f s /147fsfs LPF 0 = /147 1-D Resampling of Audio Convert CD (44.1 kHz) to DAT (48 kHz) Direct implementation Simplify LPF cascade to one LPF with 0 = /160 Impractical because 160 f s = 7.056 MHz
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7 - 9 1-D Resampling of Audio Practical implementation –Perform resampling in three stages –First two stages increase sampling rate Alternative: Linearly interpolate CD audio –Interpolation pulse is a triangle (frequency response is sinc squared) –Introduces high frequencies which will alias
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7 - 10 M x[n]x[n]xd[n]xd[n] k i is a distinct coset vector Multidimensional Downsampling Downsample by M –Input | det M | samples –Output first sample and discard others Discards data May cause aliasing
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7 - 11 Coset Vectors Indices in one fundamental tile of M –|det M| coset vectors (origin always included) –Not unique for a given M –Another choice of coset vectors for this M: { (0, 0), (0, 1), (1, 0), (1, 1) } Set of distinct coset vectors for M is unique (0,0)(1,0) (1,1)(2,1) Distinct coset vectors for M
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7 - 12 Multidimensional Upsampling Upsample by L –Input one sample –Output the sample and then | det L | - 1 zeros Adds data May cause imaging X u ( ) = X(L T ) L x[n]x[n]xu[n]xu[n]
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7 - 13 Example Upsampling Downsampling
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7 - 14 Conclusion Rational rate change –In one dimension: –In multiple dimensions: Interpolation filter in N dimensions –Passband volume is 2 N / | det L | –Baseband shape related to L T Decimation filter in N dimensions –Passband volume is 2 N / | det M | –Baseband shape related to M -T
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