[Asterisk-Users] Bounty! For help with echo cancellation code.

[echo-asterisk]

>From the CLI and during a call I want to be able to:

  *** Pulse the outgoing line and record at least 50 ms of the incoming line.

  The pulse waveform must be specifiable as a series of amplitudes
  for each 1/8000 sec time slot.  It would be best of these values
  could be read from a file specified on the CLI command line.

  Timing should be synced between the pulse and the echo so that the
  delay from the pulse to the echo can be accurately determined.

  Echo cancellation should be disabled during this operation.

  This would operate similar to the echo-training code that operates
  at the initiation of a call except that this could be done at
  any time.

  The initial pulse and any echoes can be combined and saved in a
  single channel.

  Output should go to a file and should be in a simple format that
  a program such as Audacity can read, display and play. 
   

  *** Pulse the outgoing line and record at least 50 ms of the incoming line.

  Same as above EXCEPT echo cancellation would not be disabled during
  this test and the results of the echo cancellation operations should
  be recorded and saved in a separate channel.
  

  *** Change variables used to control echo cancellation.

  Only the code in mec2.h is of interest.
   
  I will help identify the variables and modify the mec2.h code as
  needed to accomplish this goal.

  There are a lot of parameters in mec2.h that may affect the quality
  of the echo cancellation.  I want to be able to adjust them 'on the
  fly' and be able to immediately hear the results.


I am open to alternative proposals which would accomplish the same goals.

Name your price.


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Re: [Asterisk-Users] Bounty! For help with echo cancellation code.

[echo-asterisk]

On Wed, Jul 14, 2004 at 10:14:19AM -0700, Bob Knight wrote:
> [EMAIL PROTECTED] wrote:
> >>>From the CLI and during a call I want to be able to:
> >
> >  *** Pulse the outgoing line and record at least 50 ms of the incoming 
> >  line.
> >
> >  The pulse waveform must be specifiable as a series of amplitudes
> >  for each 1/8000 sec time slot.  It would be best of these values
> >  could be read from a file specified on the CLI command line.
> >
> >  Timing should be synced between the pulse and the echo so that the
> >  delay from the pulse to the echo can be accurately determined.
> >
> >  Echo cancellation should be disabled during this operation.
> >
> >  This would operate similar to the echo-training code that operates
> >  at the initiation of a call except that this could be done at
> >  any time.
> >
> >  The initial pulse and any echoes can be combined and saved in a
> >  single channel.
> >
> >  Output should go to a file and should be in a simple format that
> >  a program such as Audacity can read, display and play. 
> >   
> >
> >  *** Pulse the outgoing line and record at least 50 ms of the incoming 
> >  line.
> >
> >  Same as above EXCEPT echo cancellation would not be disabled during
> >  this test and the results of the echo cancellation operations should
> >  be recorded and saved in a separate channel.
> >  
> >
> >  *** Change variables used to control echo cancellation.
> >
> >  Only the code in mec2.h is of interest.
> >   
> >  I will help identify the variables and modify the mec2.h code as
> >  needed to accomplish this goal.
> >
> >  There are a lot of parameters in mec2.h that may affect the quality
> >  of the echo cancellation.  I want to be able to adjust them 'on the
> >  fly' and be able to immediately hear the results.
> >
> >
> >I am open to alternative proposals which would accomplish the same goals.
> >
> >Name your price.
> 
> How about being able to "see" the results real time?
> I use a package called SMAART from siasoft.com.
> It is a dual channel spectrum analyzer.
> Run the output line as your reference channel
> and the input line as your measurement channel.
> 
> You can get great info from the impulse response
> and transfer function.
> 
> You could also use this to compare different codecs.
> The impulse function will tell you how long it takes.
> The transfer function will tell you just how good a
> job it did at reconstruction the original audio.
> 

Almost 20 years ago I wrote my own digital spectrum analyzer code
which I then used to do my research.  Provided that SMAART can fully
utilize the transfer function (do convolutions etc) it would may be
useful, but spectrum analysis is not the hard part.  Controlling and
getting the data out of zaptel.o is the hard part and help with that
is what is requested in the Bounty!

echo



> -- 
> Bob Knight
> [-w] the work option
> [EMAIL PROTECTED]
> 925-449-9163
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Re: [Asterisk-Users] Spectrum Analyizer software

[echo-asterisk]

On Thu, Jul 15, 2004 at 07:41:14PM -0400, John Todd wrote:
> 
> This was an interesting post from another list 
> I'm on.  I think that with all of the echo 
> cancellation discussion happening, that someone 
> perhaps might be on the lookout for this type of 
> software, so this post was well-timed for 
> Asterisk uses.  It is perhaps the case that those 
> of you working on volume/echo/etc stuff might 
> find this useful towards your goals.

Doing spectrum analysis is not the major hurdle.

The major hurdles are:

   Extracting data from the bowels of zaptel.o

   Modifying the echo cancellation parameters in a running zaptel.o

See my previous post "Bounty!  For help with echo cancellation code".
I was serious about the "Name your price part".

[EMAIL PROTECTED]


> 
> JT
> 
> 
> >To: [snip]
> >From: [snip]
> >Date: Thu, 15 Jul 2004 06:08:33 +0200 (CEST)
> >Subject: Re: [snip] Noise masking to defeat listening devices
> >
> >
> >
> >I didn't play with signal analysis yet. But I was looking around for some
> >software and there are some interesting pieces around.
> >
> >For Linux, there is often mentioned an excellent-looking one, Baudline
> >. Not opensource but for no money.
> >
> >For Windows, there is the $99 shareware Sigview .
> >
> >A list of various digital signal processing programs is also here:
> > but many links are dead.
> >
> >
> >An interesting program is also SkySweep
> >.
> >
> >Lots of software can be found here:
> >
> >
> >libDSP  is an open-source
> >collection of various math functions for digital signal processing. A
> >highly optimized library for Fourier transformation is FFTW,
> >. These will be useful if you would like to grow
> >your own DSP software.
> >
> >
> >And there's much more. But Baudline is what I heard about the most often.
> >
> >
> >On Wed, 14 Jul 2004, [snip] wrote:
> >
> >> Check the archives.  There was a spirited discussion some time ago about
> >> PC software audio spectrum analysis.  The program I use is no longer
> >> available (adobe took it off the market), but there are s number of
> >> excellent ones out there at low cost.  Some of those provided by Amateur
> >> operators are excellent.
> >>
> > > Cheers,
> > > [snip]
> > >
> > > [snip] wrote:
> > >
> >> >
> >> >   I have also been testing a Windows application written by REI, the
> >> > ?*?ASA-2000?*? audio spectrum analyzer.  This seems to be a well 
> >> written
> >> > application and it provides a lot of good information using a standard
> > > > laptop (with sound and microphone) but I have been afraid to ask what
> >>
> >[snip]
> 
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Re: [Asterisk-Users] Current echo status?

[echo-asterisk]

On Thu, Jul 15, 2004 at 08:34:18PM -0600, Rich Adamson wrote:
> > I've been following the list for months, and I have a working Asterisk 
> > setup, but it'd be *really* useful to me at this point if someone could 
> > summarize when Asterisk has echo problems and when it doesn't.  For 
> > instance, I usually hear a far-end echo when talking on my 7940, but 
> > not when using a POTS phone plugged into a TDM400 FXO port.  It doesn't 
> > seem to matter if the call goes out over a POTS line or via NuFone; 
> > either way there's a fair bit of echo on most (but not all) calls.
> > 
> > Do different SIP phones have better echo cancellation then the Cisco 
> > 7940/60s?  How about the Polycom IP500/600s?
> > 
> > Does it matter if calls go out via POTS/T-1/PRI/VoIP?
> > 
> > The general impression that I've received is that "fast" channels 
> > (basically traditionally telephony interfaces) don't exhibit noticeable 
> > echo, but the slight delay associated with VoIP packetization unmasks 
> > existing echo.  Is that a reasonable summary?
> > 
> > We're starting to plan for a new office build-out at work, and I'd love 
> > to use Asterisk and SIP phones in the new office, but I'm not going to 
> > try to sell management on a phone system with a horrible echo problem, 
> > even if it will get fixed eventually.
> 
> For the most part, echo is less of a problem today then what it was
> a month or so ago. The source of the echo varys case by case, however
> key points that need to be addressed in _any_ installation include:
>  - * fxo pstn interface cards (eg, x100p, tdm-fxo) have to be configured
>to match the impedance of the telco standard for whatever country
>you're in. The x100p, by design, appears to be limited to a 600 ohm 
>telco impedance (US standard), and will likely generate echo when
>attached to pstn lines that are not 600 ohms. The tdm cards are
>different and can be configured to match just about any telco standard
>worldwide. (Don't know about the x101p, never seen one.)
>  - echo issues with the zap interfaces (eg, x100p, tdm, T1/E1 cards)
>rely on a software echo canceler within asterisk. If the above
>impedance match is correct, most implementations seem to have found
>parameters that minimize echo for them. However, there are still some
>in which echo is a problem and best 'guess' as of this afternoon is
>those cases appear to have something to do with undocumented internal 
>system hardware. Some folks have found swapping a motherboard for
>another with no other changes reduces echo by noticable amounts.
>That would suggest buss speed, PCI version, or something like that
>other then processor speed or RAM. There are working examples of
>300 mhz machines with no echo, 2.2 ghz machines with echo, and dual
>processor systems with echo.
>  - Digital interfaces to * (eg, T1/E1, PRI, ISDN) tend to be less prone
>to echo, however there are some implementations that still have it
>and non-technical users of those systems do complain. Technical
>users tend to tune it out.
>  - there has been very little list noise associated with echo that 
>could be honestly pinned on any sip phone.
>  - there's been a lot of opinions stated about the cause of echo on 
>the list, and at least some have no technical factual basis.
>  - there's a rather strong belief that additional echo problems exist
>within asterisk, and a group of non-programmer types are attempting
>to isolate common items in an effort to document the problem for
>those that have the programming skills to address. (That's happening
>on a non-asterisk email list.)
> 
> It seems that production systems either don't have any echo issues, or
> they have objectionable amounts. There does not seem to be anything
> in between. Its probably fair to say the majority (if no all) developers
> don't have the problem, making it most difficult to isolate and
> troubleshoot the cause.
> 
> Best guess (based only on the list noise level since Octomber 2003) is
> that probably 80% of the production * systems are near echo free, and
> 20% still have objectionable echo. Others may argue a different ratio. 
> YMMV.
> 
> Rich
> 

Well, I must be among that 20%.  I have annoying echo on PRI lines and
have tried everything available to mitigate the problem, with no
success.

I am a programmer with sufficient technical skills to attack the
problem.  HOWEVER, I am not about to spend the time it would take to
decipher zaptel.c all by myself.

I OFFER to PAY someone who has an understanding of zaptel.c to help me
with this problem.  Any takers out there???

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