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Filtering and Downsampling

Figure 10.5: Lowpass filtering followed by downsampling.
\includegraphics[scale=0.8]{eps/downsampledfilter}
Because downsampling by $ N$ will cause aliasing for any frequencies in the original signal above $ \vert\omega\vert > \pi/N$, the input signal may need to be first lowpass filtered to prevent aliasing, as shown in Fig.10.5. Suppose we implement such an anti-aliasing lowpass filter $ h(n)$ as an FIR filter of length $ M$ with a cutoff frequency $ \pi/N$. This is drawn in direct form in Fig.10.6.

Figure 10.6: Direct-form implementation of an FIR anti-aliasing lowpass filter followed by a downsampler.
\includegraphics[scale=0.8]{eps/down_FIR}

We do not need $ N-1$ out of every $ N$ filter output samples due to the $ N:1$ downsampler. To realize this savings, we can commute the downsampler through the adders inside the FIR filter to obtain the result shown in Fig.10.7. The multipliers are now running at $ 1/N$ times the sampling frequency of the input signal, $ x(n)$. This reduces the computation requirements by a factor of $ 1/N$. The downsampler outputs are called polyphase signals. This is a summed polyphase filter bank in which each ``subphase filter'' is a constant scale factor $ h(m)$.

Figure 10.7: FIR lowpass filter with downsampler commuted inside the direct-form filter.
\includegraphics[scale=0.8]{eps/down_FIR_com}

The summed polyphase signals of Fig.10.7 can be interpreted in the following ways:

  1. A ``serial to parallel conversion'' from a stream of scalar samples $ x(n)$ to a sequence of length $ M$ buffers every $ N$ samples, followed by a dot product of each buffer with $ h(0:M-1)$.

  2. The overall system is equivalent to a round-robin demultiplexor, with a different gain $ h(m)$ for each output, followed by an $ M$-sample summer which adds the ``de-interleaved'' signals together:

Figure: Demultiplex-and-sum interpretation of the polyphase signal sum of Fig.10.7.
\begin{figure}\input fig/periodicGain.pstex_t
\end{figure}

The polyphase processing in the anti-aliasing filter of Fig.10.7 is as follows:

These scaled subphase signals are summed together to form the output signal

$\displaystyle y(n) = \sum_{m=0}^{N-1} h(m)x(nN-m).
$


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``Spectral Audio Signal Processing'', by Julius O. Smith III, (October 2008 Draft).
Copyright © 2009-06-10 by Julius O. Smith III
Center for Computer Research in Music and Acoustics (CCRMA),   Stanford University
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