[arm-allstar] software update status

[Mark Guibord]

Bob,

 

The RTCM utilizes local loop back of the audio in hardware if you configure it for 'Mode 3" operation so it not a problem.

 

Mark - K7IZA

 

From: arm-allstar-bounces at hamvoip.org [mailto:arm-allstar-bounces at hamvoip.org] On Behalf Of kk6ecm
Sent: Tuesday, September 01, 2015 7:31 PM
To: ARM Allstar
Subject: Re: [arm-allstar] software update status

 

Is this an issue with usbradio on ACID (higher throughput burden) as well?

Thanks,

Bob

kk6ecm

 

Sent from iPad

 


On Sep 1, 2015, at 1:01 PM, Tim Sawyer <tisawyer at gmail.com> wrote:

David,

 

Sounds like you've found the root cause of this and have a fix is the works.

 

Excellent!

 

I think this same problem is happening with RTCM's as well. I does seem less frequent or harder to detect by ear. That may have more to do with using ACID on Intel Atom hardware than it does with RTCM's, don't know. 

 

Is your fix only to simpleUSB, or is there a chance it will help RTCM's as well? 

 

On Mon, Aug 31, 2015 at 11:24 PM, David McGough <kb4fxc at inttek.net> wrote:


Hi Bob,

The variable audio delay issue is occasionally seen on PC-computer
platforms, too, particularly with the simpleusb driver. This is NOT the
audio delay that is unavoidable due to buffering as the audio samples pass
through Asterisk. Nor is this issue related to the Asterisk jitter buffer,
etc.

Basically stated, the problem is that the input sample rate -dynamically-
varies from the output sample rate. Depending on various factors, you may
have too many samples, or too few, at any given moment. With the current
code, the main issue seen is that too many output sound frames get queued,
causing a frequently increasing in length input to output audio
delay--sometimes with delays of maybe as much as one second (really
annoying!). If there is an output buffer under-run, you hear a pop/click
due to running out of samples.

Initially, I suspected some kind of subtle bug in the simpleusb sound
code. But, after experimentation, I've been able to easily duplicate this
issue using my native ALSA stand-alone test software or even ALSA
applications like arecord and aplay, etc.

One solution to this problem is to use a dynamic buffer that effectively
re-samples the audio, gently up-sampling or down-sampling, depending on
the current trends for overrun or under-run.  I've got working software
that accomplishes this now. And, it runs full-duplex beautifully for 24+
hours with no change in audio delay and no logged clicks or pops.

Hopefully, I'll get a few days time when I'm rested to focus on completing
the integration of this code into simpleusb.  So far, I've had no time at
all to complete this.

But, hopefully soon.

73, David KB4FXC




On Mon, 31 Aug 2015, kk6ecm wrote:

> Couple of questions:
>
> 1) Is this audio delay the same as experienced with ACID?
>
> 2) What is the nature of the fix that reduces the delay?
>
> Thanks,
> Bob
> kk6ecm
>
> Sent from iPad
>
>
> > On Aug 27, 2015, at 9:33 AM, David McGough <kb4fxc at inttek.net> wrote:
> >
> >
> > Hi LaRoy,
> >
> > Progress has been slow in recent weeks, due primarily to family
> > commitments that I've got. At this point, the update is waiting on me.
> >
> > I am hopeful to minimally have a bug-fix interim release soon. The big
> > item on the list being resolution for the variable audio delay.
> >
> > I hope to have more details soon.
> >
> >
> > 73, David KB4FXC
> >
> >
> >
> >> On Thu, 27 Aug 2015, LaRoy McCann wrote:
> >>
> >> Doug,
> >> Just curious on how things are progressing with a new software update
> >> for the PI2?
> >>
> >> Thanks,
> >>
> >> LaRoy McCann, K5TW
> >>
> >>
> >>
> >>
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Tim

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