On Mar 6, 10:05 pm, "Fred Marshall" <fmarshallx@remove_the_x.acm.org>
wrote:
> "DSP-Newbie" <N...@way.invalid> wrote in message
>
> news:[email protected]..
>
>
>
> > cincy...@gmail.com wrote:
>
> >> Explicitly padding the input signal with zeros like this will work.
> >> However, note that you're essentially delaying the input signal by
> >> length(H) samples, so your output signal (even before you see any
> >> transients) will also be delayed by length(H) samples.
>
> > I am aware of that, but its is not a problem; the delay is the same for
> > all samples.
>
> >>> My problem now was, which part of Y[] should I extract and use?
> >>> After much and trial & error and some hair-pulling, I finally found a
> >>> intuitive solution that actually works:
>
> >> -snipped-
>
> >> I forgot to ask before: why do you want to discard the transients? The
> >> filter "startup" time is valid data, and occurs as your input signal
> >> gradually fills up the delay lines that feed each filter tap.
>
> > Perhaps I have not explained my problem clearly.
> > I have set the soundcard to sample at 11.025 KHz, and to report the
> > samples in blocks of 2048 samples, so I have to process about 5
> > samples/sec.
>
> > Each of these samples - after filtering, envelope detection, PLL bit
> > sampler etc... - represents a number of bits, not necessarily a *whole*
> > number of bits, which I have to cobble together block by block to
> > reconstruct the original synchronous bitstream.
>
> > Now, if I apply the algorithm as found in the book for each incoming
> > datablock, I will have a startup transient for *each* block.
>
> There's a problem you don't need to carry along. Don't do what the book
> did. Do continuous processing - if even the continous process has to wait
> between availability of data. Something like this:
>
> Get 2048 data points:
> Compute 2048 output points. Initially this will include the M-1 startup
> transient points.
> Wait for another 2048 block.
> Continue where you left off using M-1 values of the last input block still
> in the filter and the one next new input point.
> There will be no transients in the second and succeeding blocks.
>
> The code won't look all that much different from what you're using.
>
> Take a look at the circular code at:http://www.dspguru.com/sw/lib/fir_algs_1-0.c
>
> Fred

I'm amazed that it's even working at all. Or maybe he really is
keeping the filter history and doesn't realize it. I can't see how you
would get anything meaningful if the filter history was flushed after
every block; you'd have to be really lucky, I guess.

Jason