Return-Path: Received: (qmail 16621 invoked from network); 21 Mar 2001 12:36:13 -0000 Received: from unknown (HELO murphys-inbound.servers.plus.net) (212.159.14.225) by excalibur.plus.net with SMTP; 21 Mar 2001 12:36:13 -0000 Received: (qmail 5844 invoked from network); 21 Mar 2001 12:36:10 -0000 Received: from unknown (HELO post.thorcom.com) (212.172.148.70) by murphys with SMTP; 21 Mar 2001 12:36:10 -0000 Received: from majordom by post.thorcom.com with local (Exim 3.16 #2) id 14fhht-0002fy-00 for rsgb_lf_group-outgoing@blacksheep.org; Wed, 21 Mar 2001 12:27:17 +0000 X-Priority: 3 X-MSMail-Priority: Normal Received: from hestia.herts.ac.uk ([147.197.200.9]) by post.thorcom.com with esmtp (Exim 3.16 #2) id 14fhhs-0002ft-00 for rsgb_lf_group@blacksheep.org; Wed, 21 Mar 2001 12:27:16 +0000 Received: from [147.197.200.44] (helo=gemini) by hestia.herts.ac.uk with esmtp (Exim 3.16 #4) id 14fhhW-0001j0-00 for rsgb_lf_group@blacksheep.org; Wed, 21 Mar 2001 12:26:54 +0000 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2800.1106 Message-ID: <27356.200103211226@gemini> From: "James Moritz" Organization: University of Hertfordshire To: rsgb_lf_group@blacksheep.org Date: Wed, 21 Mar 2001 12:31:44 +0000 MIME-Version: 1.0 Content-Type: text/plain; charset=US-ASCII; format=flowed Content-Transfer-Encoding: 8bit Subject: LF: Soundcard calibration X-Mailer: Pegasus Mail for Win32 (v3.11) Precedence: bulk Reply-To: rsgb_lf_group@blacksheep.org X-Listname: rsgb_lf_group Sender: Dear LF group, Rik's method of producing an accurate audio tone for calibrating the soundcard is very similar to what I am doing, and I can confirm this gives an accurate and unambiguous result. I did have some problems, however - the logic level square wave (in my case 1kHz) contains many harmonics, and I found that my soundcard did not have a very good anti-aliasing filter. The result of this was that many spurious frequency components were present in the output data, which could confuse the calibration program. I overcame this by adding a simple filter - 10k in series with the divider logic output, connected to a parallel - tuned circuit consisting of a 100mH inductance in parallel with 220n + 22n capacitors. This made the output a fairly clean sine wave. You may have to fiddle with the capacitors to peak up the output, or for different audio frequencies. If you don't want to build a calibrator, there are a couple of ways of using a receiver as a calibrated audio source. If the reference oscillator output in the RX is reasonably accessible, it can be coupled with a capacitive probe wire to the RX input. If the RX is tuned to the reference frequency, the audio output will then be the nominal value within the tolerance of the reference; errors in RX tuning due to the reference will cancel out. This should work well with an RX like the RA1792, where it is easy to get at the reference output. If the reference is not available, the frequency offset can be eliminated by a differential method; Tune the RX to a stable carrier, so as to produce a fairly high frequency audio output - I used 1.3kHz. Record several minutes at 8k samples/sec, and measure the exact frequency using the Wolf -m option. Re-tune the RX on the same carrier to produce a low frequency audio output - I used 300Hz. Repeat the frequency calibration using Wolf -m. Calculate the difference between the two RX frequency settings, and the difference between the frequencies as measured by Wolf. The soundcard sample rate is then: 8000 x (difference in RX tuning setting) / (difference in measured audio frequency) This gave me almost the same results as calibrating with an accurate audio frequency. Both methods assume the receiver reference oscillator is accurate, but it's accuracy will normally be pretty good compared to the soundcard. Note, however, that both methods may be defeated by receivers using DDS synthesisers, which tune in odd-frequency steps, but display the output to the nearest Hz. Note also that the soundcard sampling rate error depends on the actual rate selected - for example, measuring 800Hz using Spectrum Lab with 8k sample rate gave me a frequency error of about +4.5Hz, but with the 11k rate, the error was about -1.6Hz. The errors are different with different types of soundcard. So using other software to measure frequency offsets will not be helpful, unless it also uses an 8k sample rate. I think Argo uses the 11k rate. I don't know about the PSK31 software. Oh joy..... Hope this is helpful, Cheers, Jim Moritz 73 de M0BMU