problems with soundmodem (userspace)

Hi,
I have several problems setting up the userspace version of soundmodem with 9k6. Everything works perfect if I only send a small set of characters (like c or m) - but sometimes a bit slow. If I enter a larger amount of text (like a message) the connection is lost after about 10 minutes of retries. Any hints what I'm doing wrong? I use a Yaesu FT-817 with Debian testing (Kernel 2.4.22, ALSA). The parameters of soundmodem are as follows:

txdelay="150" slottime="30" ppersist="30" fulldup="0" txtail="25"

As terminal I use xcall.

Is it better to use the kernelspace driver?

vy 73 de Jochen DG1PSI

From what I've seen, problems with the user-mode soundmodem driver can fall into the following categories:

- Incorrect audio levels. A receiver whose audio output is too "hot" can overdrive a PC's sound card input, clipping the waveforms and distorting the signal. Similarly, a PC whose audio output level is set too high can overdrive the rig's microphone or accessory audio input, clipping the signal or overdeviating.

The 9k6 FSK modulation is, I understand, quite "touchy" in terms of the transmit audio level. Since you're driving the modulator directly, too high an audio output level can easily over-deviate your signal, and you might drive the signal out past the receiving radio's discriminator lock range.

See if you can find someone in your area who has a deviation/modulation meter, and see whether you're pushing the transmit deviation too hard.

- Incorrect setting of the radio. For 9k6 FSK, you can't use the mic and speaker jacks. You must use an audio interface which bypasses the transmit equalization circuitry, and which gives you access to the receiver's discriminator output without re-equalization. Some radios with auxiliary audio jacks have a menu setting e.g. "1200" or "9600" - if the FT-817 has such a menu, make sure you've selected 9600.

- Sound-card resampling problems. Many modern PCs use sound interfaces based on the AC97 codecs, which have only a couple of hardware sampling rates. The soundmodem driver may be expecting to use a different sampling rate. If the soundmodem sample rate and the codec sample rates don't overlap, then somebody has to re-sample the audio data - some chipsets do this in hardware, while others have no such ability and require the CPU and drivers to do the resampling. ALSA's OSS-compatibility driver will try to do resampling, but some reports on this group seem to indicate that the resampling code has bugs in it and can result in bad samples being transmitted or received.

Also, packet transmissions in general and (I think) 9k6 in particular are often quite sensitive to RF signal quality. In particular, multipath reflections can result in enough inter-symbol interference to render the signal unreadable. Use of directional beam antennas, mounted high up in the clear, may clean up the signal enough to make the channel usable.

Another thing to check is to make sure that your radio isn't set in a power-saving mode, which leaves the receiver turned off much of the time. These modes play merry hell with packet, as the SYNC bursts are often lost.

My guess is that the kernel soundmodem driver is probably not going to be the cure for your problem... the fault lies elsewhere.

This is in fact a big problem. While soundmodem does have the resampling code and is able to use any sampling rate above a certain minimum required for the mode, some driver emulation layers interfere with this. ALSAs OSS emulation layer is a culprit here, it's also quite buggy.

I've put a soundmodem version with a rather experimental native Alsa API on http://www.baycom.org/~tom/ham/soundmodem/soundmodem-0.9.tar.gz

Tom

Hi,
thanks for your reply. I've found the problem. As you suggested it had something to do with the audio level. The soundcard-interface I use here seems to have some problems. If I directly connect the audio output of the soundcard to the data input of the TX everything works.

Thanks for your help.

vy 73 de Jochen DG1PSI

I am having the same problem with rx packets being lost so I gave this version a try. Now RX is apparently fine, but on the first TX packet the soundmodem crashes with the following output:

-cut-cut-
ALSA lib pcm_hw.c:494:(snd_pcm_hw_start) SNDRV_PCM_IOCTL_START failed: Broken pipe
sm[21164]: snd_pcm_start: Broken pipeALSA lib pcm_hw.c:494:(snd_pcm_hw_start) SNDRV_PCM_IOCTL_START failed: File descriptor in bad state sm[21164]: snd_pcm_start: File descriptor in bad statesm[21164]: audio: snd_pcm_readi: Broken pipec
-cut-cut-

I tried using something else than hw:0,0 but it either results in a config error or in a similar crash. Here is the relevant info about my Alsa setup, is there any obvious mistake in the config?

/proc/asound/cards:
0 [V8235 ]: VIA8233 - VIA 8235

                     VIA 8235 at 0x1400, irq 9

/proc/asound/devices:
0: [0- 0]: ctl
17: [0- 1]: digital audio playback
25: [0- 1]: digital audio capture
16: [0- 0]: digital audio playback
24: [0- 0]: digital audio capture
33: : timer

/proc/asound/pcm:
0-00: VIA 8235 : VIA 8235 : playback 4 : capture 1 00-01: VIA 8235 : VIA 8235 : playback 1 : capture 1

/proc/asound/version:
Advanced Linux Sound Architecture Driver Version 1.0.4rc2 (Tue Mar 30 08:19:30 2004 UTC). Compiled on Jun 12 2004 for kernel 2.6.6clarisse.

B73,
Andrea.

Homepage: http://andrea.borgia.bo.it     /    Amateur radio: IZ4FHT
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<?xml version="1.0"?>
<modem>
<configuration name="Packet">

    <channel name="Channel 0">
      <mod mode="afsk" bps="1200" f0="1200" f1="2200" diffenc="1"/>
      <demod mode="afsk" bps="1200" f0="1200" f1="2200" diffdec="1"/>
      <pkt mode="MKISS" ifname="sm0" hwaddr="IZ4FHT-1" ip="192.168.10.2" netmask="255.255.255.0" broadcast="192.168.10.255"/>
    </channel>
    <chaccess txdelay="500" slottime="100" ppersist="40" fulldup="0" txtail="10"/>
    <audio type="alsa" device="hw:0,1" halfdup="1" capturechannelmode="Left"/>
    <ptt file="/dev/ptt"/>

</configuration>
</modem>