2 Jan
2013
2 Jan
'13
5:08 a.m.
Still no luck. If I use -R for raw mode a little more data gets through (as seen by file
sizes or the Play progress meter) but it seems to be less intelligible if anything. When
I write to a file, the size is always in chunks of 16384 bytes (0x4000 or 2^14). At a
sample rate of 24000 that's 1/3 second, which is about right compared to what I hear.
Without using -R, this doesn't repeat for about a minute. That 16384 must be some
buffer size or something in the rtl_fm program, but I haven't found it yet. I think
this is supposed to repeat often enough to be continuous. Could computer speed be an
issue? I'm on a single core P4 at 3.2 GHz.
From rtl_fm -f 146970000 -f 146985000 -f 146910000 -f 146940000 scan2.dat I got this
which looks like bursts of noise as rtl_fm is scanning, still all within about 1/3
second:
http://ab1jx.webs.com/toys/dongle/scan2.dat.gif
There's a curious ripple in the baseline which doesn't quite look like power
supply ripple, more like something from some filter.
Alan
-----
Radio Astronomy - the ultimate DX
----- Original Message -----
From: Adam Nielsen <a.nielsen(a)shikadi.net>
To: Alan Corey <alancorey(a)yahoo.com>
Cc: "osmocom-sdr(a)lists.osmocom.org" <osmocom-sdr(a)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.