[Ground-station] Operation PFB

[Michelle Thompson]

Good to hear.

Edson, I believe you're working on this as well? I invited you to the
existing polyphase filter bank repository at
https://github.com/phase4ground/polyphase-filter-bank

Would you be able to manage publishing this code to this repository? We
need a reliable and knowledgeable person to maintain the polyphase filter
bank work.

This repo is where the flowgraphs from John Ackerman will be published, and
where we will be working on integrating the codebases found in the
survey. This is our home base for moving mathematical solutions into GNU
Radio and FPGAs. It also has some educational and tutorial work from fred
harris.

There are at least two other major efforts to revive, extend, and repair
the RFNoC polyphase work. Summaries of the evolution of that work can be
found here and here:

https://www.youtube.com/watch?v=DVBgLuzUlRQ
https://www.youtube.com/watch?v=WRO9VD6MYy4

If anyone knows the current status on that project, please bring us up to
date.

-Michelle W5NYV




On Tue, Jan 15, 2019 at 1:33 AM Phil Karn via Ground-Station
<ground-station at lists.openresearch.institute> wrote:

> On 11/20/18 08:56, Michelle Thompson via Ground-Station wrote:
>
> > This filter bank is in the payload. We call it a channelizer. The
> > receive bandwidth is organized into channels by this filter bank. >From
> > there, the communications are multiplexed into the time division
> > downlink DVB frames.
>
> I am very actively working on this, and of course my code will be open
> source. It's a module in my SDR package.
>
> I use fast convolution throughout my package for filtering and
> correlation. The channelizer I'm working on extends this mechanism.
>
> There are two main forms of fast convolution; I use the technique called
> "overlap and scrap". If the data blocksize is 'L' and the length of your
> filter impulse response is 'M' samples, then you use a FFT length N = L
> + M - 1. Each FFT operates on L new I/Q samples and M-1 I/Q samples from
> the previous block. Then you multiply the transform by the transform of
> your desired filter's impulse response, perform the inverse FFT, and
> take only the last L samples from the output. Overlapping and scrapping
> is necessary because the FFT performs circular convolution and we want
> linear convolution when doing filtering. This method keeps the tail of
> the convolved result from "wrapping around" and messing up the start of
> the block.
>
> I also decimate with this method. To downsample from 192 kHz to 48 kHz,
> I simply use an IFFT of length N/4 that uses only the first N/4
> frequency bins from the forward FFT. I still have to multiply by the
> filter response in the frequency domain, and I have ensure that the
> impulse response spectrum is essentially zero through all the high
> frequency bins I'm ignoring or I will get unwanted aliasing. I use a
> Kaiser window with a fairly large beta factor to give low sidelobes at
> the expense of a wide main lobe. But since I'm only using a fraction of
> the 48 kHz bandwidth for communications-quality audio, this is not a
> problem.
>
> Now here's the really fun part. Not only will the shorter IFFT execute
> faster than the bigger forward FFT, BUT MANY INVERSE FFTs CAN SHARE ONE
> FORWARD FFT. This is key to making an efficient channelizer.
>
> With the right values of L and M, you can turn this into a versatile
> channelizer by simply selecting different frequency bins from the
> forward transform for each inverse transform. Each bin will be
> Input_Samprate/N Hz wide, but your frequency shift must also be an
> integral number of cycles within L samples (as well as N). So by making
> L = M, you can shift by any even number of bins.
>
> I typically make L = M = 20 milliseconds at whatever sample rate I'm
> using (usually some multiple of 48 kHz). At 12.288 MHz, that's a forward
> FFT of N=491,520 samples. 20 milliseconds is 50 Hz, so I can shift by
> any multiple of 2*50 = 100 Hz. Each channel can be independently
> centered on any multiple of 100 Hz with an arbitrary bandwidth in
> increments of 100 Hz (with allowance for broadening from windowing.)
>
> I can certainly switch to a smaller FFT with a correspondingly higher
> channel roster, but this FFT is fast enough on my CPU so I haven't tried
> that yet.
>
> This ability to place channels on fractions of their bandwidths will
> come in handy when I channelize the entire 2m band and demodulate every
> FM signal at the same time. 2m has a convoluted band plan with 20 kHz
> channels below 146 MHz, 15 kHz above, and many oddball local variations.
>
> I don't know if a conventional polyphase filter bank can handle an
> irregular bandplan, but I know I can.
>
> My HackRF is now sampling at 12.288 MHz (256x 48 kHz) and generating a
> 200 Mb/s I/Q multicast stream on my home network. Amazingly, with a
> little tuning my switches and hosts handle this sustained data rate
> without loss.
>
> The program that handles the HackRF USB stream and generates the
> multicast RTP stream performs analog AGC, DC removal and I/Q gain and
> phase balance. It's running on an old 4-core 1.66 GHz Atom CPU, with two
> main threads consuming 63% and 23% of a core.
>
> I'm processing this stream on a 4-core (8 with hyperthreading) 2.4 GHz
> Xeon that's probably 10 years old now. The thread that processes the I/Q
> multicast stream and does the forward FFT uses 68% of one CPU and 27% of
> another. (I'm using FFTW3 with two threads, and I've run fftwf-wisdom on
> my transform sizes. 491,520 has no factors greater than 5, so it's
> fast.) The rest of the receiver runs at 48 kHz, so naturally those
> threads take up much less CPU (2-3% to demodulate FM by brute force with
> arctangent, plus a squelch driven by a SNR estimator, and other overhead
> functions).
>
> My goal in all this is to build a proof of concept for how I think a
> modern digital amateur radio satellite should work. It should have a
> single high speed downlink (for which DVB-S2 with LDPC is ideal)
> relaying a multiplexed stream of uplink digital voice and data from a
> bank of analog FM and low speed digital receivers. Each analog FM
> channel is demodulated and encoded with Opus, which sounds pretty good
> on voice at its lowest rate of 6 kb/s. Digital uplinks would be simply
> turned around and retransmitted as-is; presumably these would also be
> Opus-compressed digital voice but they could also be non-voice data
> streams.
>
> 73, Phil
>
>
>
>
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