[aprssig] TNC vs AGWPE

[Andrew Rich]

Thanks Steve !

Good notes

Yes I checked the level and did try dual channel and noticed that the mic
was different to "line in"

It is a matter of frustartion to me that this laptop has just mic in.

We bought an external USB which seems to work. for another setup.

Yep - had checked and tried the mic booster.

I even used a sound recodring program where I could see the samples !

I did my homework just got disappointing results.

Cheers Andy

-----------------------------------------
Andrew Rich
Amateur radio callsign VK4TEC
email: vk4tec at tech-software.net <mailto:vk4tec at tech-software.net>
web: http://www.tech-software.net
Brisbane AUSTRALIA



-----Original Message-----
From: Stephen H. Smith [mailto:wa8lmf2 at aol.com]
Sent: Wednesday, 19 April 2006 1:43 AM
To: TAPR APRS Mailing List
Cc: vk4tec at tech-software.net
Subject: Re: [aprssig] TNC vs AGWPE


kc5zrq at gmail.com wrote:
> Don't use the "9600" out.  As far as I know, the ISS uses 1200 baud.
>

1)     The so-called "9600 out" is not a data output.  It's actually
non-squelched non-de-emhasized AUDIO out,  directly from the receiver's
FM discriminator.    (This is the kind of audio connection you MUST HAVE
for connecting external TNCs running at 9600 baud, although it is also
usable for TNCs running at 1200.)

The "9600" output is usually about 50millivolts which will overload the
typical PC "mic input" and cause severe distortion unless you use about
a 5:1 or 10:1 attenuator pad.

Further, many PC mic input jacks have 3-5 volts DC on them to power
active electret external mics.    The proper way to couple the radio's
audio into the PC is to first use a 1:1 turns ratio audio transformer in
order to avoid a common ground between the radio and the computer.  (The
typical 600:600 ohm transformer used in telephone devices like modems,
answering machines, etc. is ideal).   The secondary side of the
transformer should be connected across a voltage divider network of two
resistors in series.  The one closer to ground should be around 1K while
the upper one typically will be somewhere between 4.7K and 10K.
Finally connect the center point of the two resistors, to the PC audio
input using something like a .1 to .5 uF series capacitor to block any
DC that may be present on the mic input.

Note that the PC mic input is SINGLE CHANNEL even though it uses a
stereo TIP-RING-SLEEVE  (3-conductor) mini-plug.   Normally the TIP
carries +5 VDC power WHILE the ring carries AUDIO.  The ring may or may
not also have 3-5 VDC on it for use with mics that carry DC and audio on
the same conductor.




2)     Once you have your audio network in place, click the AGW icon in
the system tray and choose "Soundcard Tuning Aid".  One the resulting
screen, click the radio button for  " Oscilloscope Sine Wave".   Watch
the display as packet bursts come in. You should see a clean sinewave
display here.   Use the Windows RECORD mixer (not the default PLAYBACK
mixer that you get initially when you click the speaker icon in the
tray) to adjust the audio level for a clean sine wave.  Note that you
will see white-noise random grass between bursts since this is
non-squelched audio.  It's perfectly normal for this noise to clip on
peaks.   The goal is that the sine wave DURING packet bursts be smooth
and rounded with no flat topping.

Note that the options in the Windows mixer for the mic input channel on
many sound cards includes a checkbox for a "+20 dB Mic Booster". This is
for the benefit of low-output non-amplified mics (i.e. passive dynamic
mics) instead of electret ones.  You definitely want this box UNCHECKED.




3)     Most modern PCs, both laptop and desktop no longer have dedicated
sound card hardware in them.  The built-in "AC97 Compatible" sound
systems are basically an A-D converter and nothing else.  The "heavy
lifting"  precisely-timed sampling and  processing of sound that was
traditionally done by a dedicated processor,  RAM and accurate clock on
the sound card is now done by the main CPU of the computer.


The CPU clock usually isn't as accurate as the one on a sound card.
Further, the CPU is sharing it's available pool of clock cycles between
a varying number of other processes with varying interrupt response
times (latencies).  The result is that the sampling rate is frequently
far off  of the desired 11,025 or 8,000 samples/sec  that ham soundcard
apps expect.   Further  IT VARIES depending on how many other programs
are running at the same time!    There are several utilities that will
measure the actual sampling rate of the sound card.


Unfortunately, unlike some ham sound card programs such as mmSSTV and
MixW, AGW has no provision for entering corrections for the sampling
rate errors.   If the sample rate is severely off,  your only recourse
is to try a different sound card --  in the case of a  laptop  this will
mean either a PCMCIA -card based sound system,  or an external
USB-connected one.


Ironically, the older Pentium I and Pentium II laptops often had far
superior sound systems based on dedicated Soundblaster, ESS, or Crystal
Audio chip sets just like add-on  PCI-card sound systems in desk top
PCs.    Today's hotrod P4 or Centrino laptops usually have the
far-inferior "brain-dead" host-based AC97 sound systems. But AC97
systems reduce parts count, power consumption and are CHEAP CHEAP.    I
have a couple of 10-year-old Dell 3000 Pentium I  200MHz MMX - based
laptops that I keep around exclusively for ham soundcard operating with
mmSSTV, AGW, MixW and Echolink precisely because their sound systems are
superior to my newer machines.   Further these "classic" machines have
the audio LINE input in addition to the MIC input which is far superior
for ham sound card applications.     With AGW, the true stereo line
input actually allows you to create a DUAL-PORT tnc similar to a
Kantronics KAM or 9624, with one radio connected to the left channel and
a second radio on the right channel.







Stephen H. Smith    wa8lmf (at) aol.com
EchoLink Node:      14400    [Think bottom of the 2M band]
Home Page:          http://wa8lmf.com


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