amohar

Hello everyone, I'm trying to understand the demodulation theory and thus understand the GnuRadio graphs implementing software demodulation. I'm still a bit stuck on the bare basics, so here are my questions: 1. I understand that the rational resampler can be used to lower the number of received samples by using the decimation parameter. This is helpful for speeding up the software operations while still keeping the quality. Low-level filter also has the decimation parameter, why isn't that one usually used for decimation? In what case would we use filter's decimation? 2. I understand that the AM demod (for instance) usually has two steps, rectifying the signal to baseband signal and then applying low-pass filter to get rid of unwanted high frequencies (leftovers from the modulation) and get a nice wave. My question is why do we need the low-pass filter before the AM demod (in fact WBFM as well)? Some say that it is for band selection, but shouldn't it then be the band-pass filter? 3. Is there any good information about Frequency xlating FIR filter? I know it can use firdes Python filters, but still haven't got to grips with them. What are the Taps (I know this should be the place for firdes Python call to low_pass_2, but what does it actually mean)? I guess the center frequency is the one I want to tune into... Is this the same frequency as set on SDR input source? Why is this center frequency missing in the low-pass filter block? If this frequency is translated to 0Hz (as docs say), do I still get my left channel? Just by writing these questions down I've made some things clearer to myself. I would still be very grateful if anyone would clarity on these things even more. Also, although I've read anything and everything online that I could find regarding this topic, I would appreciate any resources that you might think of as helpful. Thank you for any info, Andrej

You are right, I proved that with multiple failed attempts to make it work on my VM. As I said, booting from a burned Live DVD works, but I still sometimes get underruns and then the system freezes. I usually have to restart my computer. Luckily I went forward to fiddle with some digital signals (wireless weather station I have at home), so currently there's no need for me to dual-boot (I track the data transforms with the GUI elements which work fine in both VM and when booting the live DVD). I will have to make it happen soon, though. I'll send more info when I go around to do it (if it worked or not). Thanks for all the help!

Hey! Thank you for your response. Although the problem in the end turned up to be something entirely different, the video you posted was very helpful. It got me to using a simple single frequency sound and osciloscope for testing my problems. Once I get GnuRadio to run flawlessly I'll definitely go through the whole series. The problem seems to be running the GnuRadio Live DVD in the VmWare Player. When I was using a single frequency (1kHz), everything worked. When I shifted for 100Hz, I got underflow. That seemed strange because people in videos can change frequencies with sliders and without changing the sample rate. Also, having a plain simple variable to set the frequency (1kHz) worked while having a slider set the exact same frequency (1kHz) didn't. Not to mention adding a FFT block, which requires quite some computation power. So the reason seems to be too slow simulation capabilities on a VM (or should I say computation power that my VM is currently not capable of doing). I should mention that I tried to remedy this by adding another vCPU (making VM a dual core VM) but it hasn't helped. I burned the DVD and booted from there. The examples kinda worked, but not fully. The new problem with booting from a live DVD is the system freezing (I can move my mouse, the sound is on, but all the windows are blocked and can't be closed), even when the throttle blocks are added. So the current problems are how to make VmWare work faster or better and / or how to make Live DVD work without freezing. Any new info greatly appreciated (as always)!

Hello everyone, I'm writing my first post in hope of shedding some light on rational resampling in FM radio. My idea was to get started with something easy to get me into the SDR programming and I chose FM radio. I have watched HAK5 SDR episode where the FM radio is built. I have also found similar graph on instructables (links below). I have tried to duplicate their graphs (with my own working frequencies), but without much success. My finished graphs produce loud cracking noise in my headphones. What I noticed, though, is what looks to me a strange discrepancy in the rational resampling and sample rate settings. Now, I'm quite sure I have some basic concept wrong, because it obviously works for both of them. Here's what is strange: 1. Sample rate starts at 2M at RTL-SDR Source 2. It is resampled with interplation 1 and decimation 4, which should produce the sample rate of 500k ("2M * 1/4", if I undestand correctly) 3. Low Pass Filter has also Sample Rate 2M (Shouldn't it be 500k?) 4. WBFM Receive has Quadrature rate of 500k and Decimation 1 5. It is resampled again with Interpolation 48 nad Decimation 500. If we take 2M sample rate from Low Pass Filter, it should produce 2M * (48 / 500) = 192 kHz. But if we take 500k sample rate, it should produce 48kHz, which is also the setting of the audio sink. The problem is, if I put 500k as the Sample Rate of Low Pass Filter, instead of cracking noise I hear only silence. So my understanding is probably very much off. Can anyone shed some light on these problems? Is there another problem present, which is not tied to the GNURadio Companion and resampling? Here are the links: HAK5 episode I watched: Instructables: http://www.instructables.com/id/RTL-SDR-FM-radio-receiver-with-GNU-Radio-Companion/?ALLSTEPS Graph from instructables similar to HAK5's: http://cdn.instructables.com/FFO/PKAD/HNE45JZQ/FFOPKADHNE45JZQ.LARGE.jpg Rational resampling basics I read: http://www.dspguru.com/dsp/faqs/multirate/resampling Any help will be greatly appreciated.