Trying to use rtl_fm, etc

Alan Corey alancorey at yahoo.com
Tue Jan 1 04:15:22 UTC 2013


I've tried matching the sample rates, it doesn't seem to make much difference. I just didn't happen to have saved the text from one of those.

I've also tried the -r option, but I didn't hear anything and I wasn't sure it was sending "raw" audio and not I/Q signals.  It acted better, aside from that: the progress meter in Play was responding constantly like it was getting more audio and the file sizes were more like what I'd expect.  With the bursting, I get 16k every minute, which is way too short.

I wrote some C to import one of the files and export as ascii, then make a Gnuplot file to plot it.  There's a sample at:
http://ab1jx.webs.com/toys/dongle/wfm2_dat3.gif (The X axis labels are data point numbers.)
It looks like audio, but I see something in the output about a 6 millisecond sample buffer. That's possibly how much sound I get, and this sample is from the local NPR station so I don't know what they were doing at that instant, music or voice.  I also haven't tried plotting I/Q output so I don't know what that looks like.

Yes, my sound works and playing a wav file with Play (Sox) works.  I normally work at 8000 samples/second mono.  No Linux sound to get in the way here (OpenBSD). From reading, it doesn't make sense to have the RF sampling rate the same as the audio sampling rate (I think) but that's what it defaults to.


 OK, it helps to be reassured that somebody actually uses this and has it working.  I'll mess with sampling rates and raw mode.  Some real documentation for these programs would be a help.

  Alan

-----
Radio Astronomy - the ultimate DX



----- Original Message -----
> From: Adam Nielsen <a.nielsen at shikadi.net>
> To: Alan Corey <alancorey at yahoo.com>
> Cc: "osmocom-sdr at lists.osmocom.org" <osmocom-sdr at lists.osmocom.org>
> Sent: Monday, December 31, 2012 1:28 AM
> Subject: Re: Trying to use rtl_fm, etc
> 
>>  With rtl_fm, I get a tiny burst of audio about once a minute.
>> 
>>  A fresh run:
>>  freebie# rtl_fm -N -f 162550000 - | play -t raw -r 32k -e signed-integer -b 
> 16
>>  -c 1 -V 4 -
> 
> Just FYI, you can see here that:
> 
>>  Output at 24000 Hz.
> 
> However you have told 'play' to play the audio at a sampling rate of 
> 32kHz, even though the audio data is only arriving at 24kHz.  So you will get 
> stuttering as the audio buffer keeps running out and waiting for more data to 
> arrive.
> 
> For me (under Linux), I get best results using the -r option to rtl_fm to set 
> the output audio sampling rate to 48kHz, then tell play to play at 48kHz too.  
> This way my system doesn't have to resample it to 48kHz before it can mix 
> the stream into the system-wide audio output.
> 
> Note that the -s option sets the signal bandwidth and -r sets the output audio 
> sampling rate.  A lot of people misunderstand the purpose of the -s option, 
> however you shouldn't need it unless you are trying to receive data signals. 
> -W and -N set -s to the correct values for voice transmissions.
> 
> I would also suggest playing a .wav file with the same 'play' options 
> just to make sure your system can play mono audio at low sampling rates.  I know 
> my sound card drivers won't (possibly because I am using a SPDIF connection 
> to an external amplifier) so I need the Linux audio system to upmix it to 48kHz 
> 16-bit stereo or I won't hear anything.
> 
> Cheers,
> Adam.
> 
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