--- In AudioBox@yahoogroups.com, "jdow" <jdow@...> wrote:
>
> If all the audience is more than 20 ms away from the sound source
> the 10 ms minimum delay of SoundMan would mean nothing. You do want
> the sound from the speakers to come up just as or a touch after the
> direct sound reaches the listener, don't you? Isn't the speed of
> sound on the (very) rough order of 1 foot per ms? So at -20'
> difference between the sound arrival from a speaker location and
the
> direct sound arrival puts you in a safe ballpark.
>
> {^_-}   <- sometimes she gets obnoxious, doesn't she?
>

If the routing was for reinforcement only perhaps so, but when you
get into real-world applications where performers have to sing/play
along to monitor feed, well, I think the problem is obvious. Some
musicians claim that as little as 4 ms delay throws off their
performance, but that could be braggadocio (or an excuse). ;^>

Essentially, it would seem that SoundMan would be more of an SFX
alternative then for an AudioBox. My point was that having a fully
developed UI was important for making SoundMan successful which is
something that I certainly would like to see.

Bruce Ellman

From: "Richard B. Ingraham" <rbingraham@...>
>> From: AudioBox@yahoogroups.com
>
>> They have vastly different form factors. And the AB does not
>> need the external plugin boxes for its large number of
>> channels - or does it?
>>
>
> No it doesn't.  You can have 64 analog inputs and outputs if you want to
> spend the money.
>
>
>> 64 samples on input and output -> maybe 7 to 9 ms turn around
>> with the right audio facility figuring three or four buffer
>> pipelines for safety.
>>
>
> And that is too high for most live audio applications, plus you are
> talking about best case scenario performance, are you not?  I would
> expect most CPUs to choke fast with that low of latency, and not provide
> all that many channels of I/O.  But maybe I'm wrong?
>
>
>> Need to use almost the same amount of CPU regardless if you
>> approach the CPU bound condition. The FIFO code uses very
>> little time. The large time is in the audio processing
>> depending on how much of that is done. That's what the three
>> or four buffer pipeline is all about.
>>
>> It isn't instantaneous. It also isn't hundreds of
>> milliseconds, either.
>>
>
> I don't know what most of that means.... nor do I care to be completely
> honest...  :-))
>
> But maybe I am missing something?
>
> My thought is this....  if SoundMan is limited because of current sound
> card drivers to be mostly used as a playback device, then it would seem
> to me, that you might as well just crank up the latency on the sound
> card's drivers, reduce the load on the CPU, and be able to produce more
> channels of audio for a given amount CPU and other system resources.

Increasing the latency does not reduce CPU load materially. Arguably it
increases the CPU load if it is done wrong. Reducing the latency to
less than three or four packets does not allow for times the machine
decides to do its own thing. But more than 4 packets starts to get
silly and increases the computing challenge "slightly". The real problem
is pushing the samples in from the source, through filters, delays,
level shifts, matrix connection calculations, and on out the output.

(And live is less work than clip because there is no file system
overhead involved.)
{^_^}

> At least in my mind, for playback only latency of 25 - 40 ms is probably
> acceptable from the time you hit "GO" until a sound is produced.  Unless
> you are maybe trying to play SoundMan like a keyboard workstation.
>
>
>> An input's an input regardless of whether it reads off
>> firewire or off disk. (Actually live input off firewire is
>> probably less overhead.)
>
> So you are saying that routing live audio is actually less overhead than
> same number of hard disk playback channels?  Assuming the same latency?
> But obviously if your doing playback only and you don't need such low
> latency, you could crank up the latency and get more channels for the
> CPu buck, yes?
>
>> Um, I have the initial piece of E-Show2 written. It works. So
>> we're setup to do SoundMan and ShowMan together. Loren's just
>> horridly backed up from day job work.
>
> So is E-Show2 to include the RTP-MIDI stuff?  How about the fix for the
> AB-SCSI-MIDI tool for use with USB-SCSI interfaces?
>
> Thanks,
>
> Richard B. Ingraham
> RBI Computers and Audio
> http://www.rbicompaudio.20m.com/
> rbI@...
>
>
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> 10/13/2006
>
>
>
>
> To unsubscribe from this list
> mailto:AudioBox-unsubscribe@yahoogroups.com
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>
>
>

From: "Richard B. Ingraham" <rbingraham@...>
>> From: AudioBox@yahoogroups.com
>>
>> How little does he really need? That is a market
>> differentiator between AudioBox and SoundMan, though.
>
> My point exactly......  so why bother trying to route dozens of channels
> of live audio, if the latency is going to be too high for most
> applications.
>
> Although I guess it could be useful for things like building paging and
> such.... but is that really a potential market for SoundMan?  I would
> doubt it.... but maybe I'm missing the point?

If all the audience is more than 20 ms away from the sound source
the 10 ms minimum delay of SoundMan would mean nothing. You do want
the sound from the speakers to come up just as or a touch after the
direct sound reaches the listener, don't you? Isn't the speed of
sound on the (very) rough order of 1 foot per ms? So at -20'
difference between the sound arrival from a speaker location and the
direct sound arrival puts you in a safe ballpark.

{^_-}   <- sometimes she gets obnoxious, doesn't she?

> -----Original Message-----
> From: AudioBox@yahoogroups.com
> [mailto:AudioBox@yahoogroups.com] On Behalf Of jdow
> Sent: Saturday, October 14, 2006 6:49 PM
> To: AudioBox@yahoogroups.com
> Subject: Re: [AudioBox] Re: FW: another wild Richard idea....
> just to drive you crazy!
>
>
> How little does he really need? That is a market
> differentiator between AudioBox and SoundMan, though.

My point exactly......  so why bother trying to route dozens of channels
of live audio, if the latency is going to be too high for most
applications.

Although I guess it could be useful for things like building paging and
such.... but is that really a potential market for SoundMan?  I would
doubt it.... but maybe I'm missing the point?

Richard B. Ingraham
RBI Computers and Audio
http://www.rbicompaudio.20m.com/
rbI@...


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> -----Original Message-----
> From: AudioBox@yahoogroups.com
> [mailto:AudioBox@yahoogroups.com] On Behalf Of jdow
> Sent: Saturday, October 14, 2006 3:20 AM
> To: AudioBox@yahoogroups.com
> Subject: Re: [AudioBox] FW: another wild Richard idea....
> just to drive you crazy!
>

> They have vastly different form factors. And the AB does not
> need the external plugin boxes for its large number of
> channels - or does it?
>

No it doesn't.  You can have 64 analog inputs and outputs if you want to
spend the money.


> 64 samples on input and output -> maybe 7 to 9 ms turn around
> with the right audio facility figuring three or four buffer
> pipelines for safety.
>

And that is too high for most live audio applications, plus you are
talking about best case scenario performance, are you not?  I would
expect most CPUs to choke fast with that low of latency, and not provide
all that many channels of I/O.  But maybe I'm wrong?


> Need to use almost the same amount of CPU regardless if you
> approach the CPU bound condition. The FIFO code uses very
> little time. The large time is in the audio processing
> depending on how much of that is done. That's what the three
> or four buffer pipeline is all about.
>
> It isn't instantaneous. It also isn't hundreds of
> milliseconds, either.
>

I don't know what most of that means.... nor do I care to be completely
honest...  :-))

But maybe I am missing something?

My thought is this....  if SoundMan is limited because of current sound
card drivers to be mostly used as a playback device, then it would seem
to me, that you might as well just crank up the latency on the sound
card's drivers, reduce the load on the CPU, and be able to produce more
channels of audio for a given amount CPU and other system resources.

At least in my mind, for playback only latency of 25 - 40 ms is probably
acceptable from the time you hit "GO" until a sound is produced.  Unless
you are maybe trying to play SoundMan like a keyboard workstation.


> An input's an input regardless of whether it reads off
> firewire or off disk. (Actually live input off firewire is
> probably less overhead.)

So you are saying that routing live audio is actually less overhead than
same number of hard disk playback channels?  Assuming the same latency?
But obviously if your doing playback only and you don't need such low
latency, you could crank up the latency and get more channels for the
CPu buck, yes?

> Um, I have the initial piece of E-Show2 written. It works. So
> we're setup to do SoundMan and ShowMan together. Loren's just
> horridly backed up from day job work.

So is E-Show2 to include the RTP-MIDI stuff?  How about the fix for the
AB-SCSI-MIDI tool for use with USB-SCSI interfaces?

Thanks,

Richard B. Ingraham
RBI Computers and Audio
http://www.rbicompaudio.20m.com/
rbI@...


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> -----Original Message-----
> From: AudioBox@yahoogroups.com
> [mailto:AudioBox@yahoogroups.com] On Behalf Of jdow
> Sent: Saturday, October 14, 2006 3:21 AM
> To: AudioBox@yahoogroups.com
> Subject: Re: [AudioBox] Re: FW: another wild Richard idea....
> just to drive you crazy!

> Precisely how little?
> {^_-}

I would never want to put a sound system together that has more than 5ms
of total latency through out the entire system.  And this means all
devices from the time the sound hits the mic, until it comes out the
speakers. (obviously longer if I have put in more delay time
intentionally....)

In my very rudimentary experiments many years ago, I couldn't notice any
shift in image up to 5ms.  But I could definitely hear image shift by
10ms, and sometime less.... but I wasn't sure if that was just my eyes
telling me that there was a shift, or if I was actually hearing it.
Anyway I know I can hear 10ms.

So since then I've always tried to keep it under 5ms total.  This is not
really that hard to do actually.  I regularly run shows where my all or
almost all of my audio passes through both a Yamaha digital mixer and an
audiobox, or other DSP unit.  Even with those 2 conversions (A to D, and
D to A) it's still under 5ms total.

Now if I was to use SoundMan rather than a AB, that would crank the
total latency up close to at least 10ms, or more.  That's too much
system delay, as it will start to cause a noticeable shift in image at
that point.  Granted, I'll often end up adding more delay in, to create
some precedence to the actors actual voice, but I want to be in charge
of that, and not restricted by it.  Plus I don't put that on all of the
channels.

Richard B. Ingraham
RBI Computers and Audio
http://www.rbicompaudio.20m.com/
rbI@...


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From: "Charlie Richmond" <charlier@...>

> On Sat, 14 Oct 2006, jdow wrote:
>
>>> 1.  I figured that way you wouldn't be trying to market two separate
>>> products that are a direct competition to each other.  The AB64 would
>>> still have the ability to route many more inputs around than SoundMan
>>> server.  But they could both have lots of outputs.
>>
>> They have vastly different form factors. And the AB does not need
>> the external plugin boxes for its large number of channels - or
>> does it?
>
> No, it doesn't unless you use Cobranet or Ethersound and external converters 
-
> but then, that can be more convenient because the analog lines don't have to
be
> run so far.
>
>> 64 samples on input and output -> maybe 7 to 9 ms turn around with
>> the right audio facility figuring three or four buffer pipelines for
>> safety.
>
> This is the problem of course - that is many times the AB's latency.
>
>> Um, I have the initial piece of E-Show2 written. It works. So we're
>> setup to do SoundMan and ShowMan together. Loren's just horridly
>> backed up from day job work.
>
> Good and bad, as usual!
>
>> {^_^}   (And I just packed up with Matrox DSX video card work.)
>
> What does this mean?  You will have more time now???  ;-)

No, I packed up my job cue as tight as it goes. This is going to go for
"months" most likely.
{^_^}

How little does he really need? That is a market differentiator
between AudioBox and SoundMan, though.

{^_-}
----- Original Message -----
From: "Charlie Richmond" <charlier@...>


> On Sat, 14 Oct 2006, jdow wrote:
>
>>> Besides, I don't think SoundMan will replace the AB64 for those of us who
>>> like to use the AB as their main drive device for matrix, eq, delay, etc and
>>> need to pass live audio with very little (and predictable) latency.
>>
>> Precisely how little?
>
> 1.4mS, analog in to analog out.
>
> C

On Sat, 14 Oct 2006, jdow wrote:

>> 1.  I figured that way you wouldn't be trying to market two separate
>> products that are a direct competition to each other.  The AB64 would
>> still have the ability to route many more inputs around than SoundMan
>> server.  But they could both have lots of outputs.
>
> They have vastly different form factors. And the AB does not need
> the external plugin boxes for its large number of channels - or
> does it?

No, it doesn't unless you use Cobranet or Ethersound and external converters  -
but then, that can be more convenient because the analog lines don't have to be
run so far.

> 64 samples on input and output -> maybe 7 to 9 ms turn around with
> the right audio facility figuring three or four buffer pipelines for
> safety.

This is the problem of course - that is many times the AB's latency.

> Um, I have the initial piece of E-Show2 written. It works. So we're
> setup to do SoundMan and ShowMan together. Loren's just horridly
> backed up from day job work.

Good and bad, as usual!

> {^_^}   (And I just packed up with Matrox DSX video card work.)

What does this mean?  You will have more time now???  ;-)

C

On Sat, 14 Oct 2006, jdow wrote:

>> Besides, I don't think SoundMan will replace the AB64 for those of us who
>> like to use the AB as their main drive device for matrix, eq, delay, etc and
>> need to pass live audio with very little (and predictable) latency.
>
> Precisely how little?

1.4mS, analog in to analog out.

C

From: "Richard B. Ingraham" <rbingraham@...>
>> From: AudioBox@yahoogroups.com
>>
>> If you throw enough hardware it is estimated that SoundMan
>> would poop out well above a piddly eight in/eight out
>> scenario. Loren tests routinely with MOTU 828 and a bank of
>> three expanders.
>>
>> The processing speed of the attached computer is the limiting factor.
>>
>> {^_^}
>
> Of course, but I only suggested limiting the number of live inputs....
> not the number of outputs, because:
>
> 1.  I figured that way you wouldn't be trying to market two separate
> products that are a direct competition to each other.  The AB64 would
> still have the ability to route many more inputs around than SoundMan
> server.  But they could both have lots of outputs.

They have vastly different form factors. And the AB does not need
the external plugin boxes for its large number of channels - or
does it?

> 2.  The latency involved in passing audio through the sound card to the
> CPU and back again is just too large for most applications anyway.  So
> unless you are just routing effects processors, or some other playback
> device, etc...  I don't think live audio routing via software is ready
> for prime time yet.  Even with some better sound cards that can get
> really impressive latency (like RME for example) there is still about
> twice as much latency (at a minimum) as most hardware digital mixers or
> DSP boxes.  For some applications that is probably still good enough,
> but not for any live sound, ala wireless mic routing, or orchestra mics,
> etc...  depends a lot on the market I guess.

64 samples on input and output -> maybe 7 to 9 ms turn around with
the right audio facility figuring three or four buffer pipelines for
safety.

> 3. Routing live audio through takes a big CPU hit.  Then the lower your
> latency, the more CPU you need to use. (obviously I know you and Loren
> know this... I'm just giving you the reasons behind my suggestion)

Need to use almost the same amount of CPU regardless if you approach
the CPU bound condition. The FIFO code uses very little time. The
large time is in the audio processing depending on how much of that
is done. That's what the three or four buffer pipeline is all about.

It isn't instantaneous. It also isn't hundreds of milliseconds, either.

> 4. In comparison, I wouldn't want to limit the number of outputs,
> because I see it as mostly useful as a playback device.  Plus I don't
> think having more outputs chews up anywhere near as much CPU as does
> routing live inputs.  Unless maybe you are comparing an output with
> delay and EQ, to an input with no processing.  And has always been the
> case in theatre, we are usually dealing with systems that have more
> outputs than inputs (at least when you are talking about playback only
> shows) or at just as many outputs as inputs.

An input's an input regardless of whether it reads off firewire or
off disk. (Actually live input off firewire is probably less overhead.)

> I think you get my points..   :-))
>
> My main point however wasn't any of this.... it was to make a couple of
> small changes to the Server/Audio Engine that as far as I can tell is
> essentially done, and get it out the door.  :-))

Um, I have the initial piece of E-Show2 written. It works. So we're
setup to do SoundMan and ShowMan together. Loren's just horridly
backed up from day job work.

{^_^}   (And I just packed up with Matrox DSX video card work.)

From: "Bruce Ellman" <bruce@...>

> Richard,
>
> I think your suggestion to get SoundMan out there sooner rather then later
> as an SFX alternative is a good one, but I think it's the UI that will help
> get it the market penetration into SFX's near monopoly. SoundMan/AB64 is far
> more powerful then SFX, but you still need to have an interface that is more
> user friendly then its current incarnation to overcome market inertia.
>
> Besides, I don't think SoundMan will replace the AB64 for those of us who
> like to use the AB as their main drive device for matrix, eq, delay, etc and
> need to pass live audio with very little (and predictable) latency.

Precisely how little?
{^_-}

Richard,

I think your suggestion to get SoundMan out there sooner rather then later
as an SFX alternative is a good one, but I think it's the UI that will help
get it the market penetration into SFX's near monopoly. SoundMan/AB64 is far
more powerful then SFX, but you still need to have an interface that is more
user friendly then its current incarnation to overcome market inertia.

Besides, I don't think SoundMan will replace the AB64 for those of us who
like to use the AB as their main drive device for matrix, eq, delay, etc and
need to pass live audio with very little (and predictable) latency.

Best,
Bruce


     -- Bruce Ellman
        bruce@...




[Non-text portions of this message have been removed]

> -----Original Message-----
> From: AudioBox@yahoogroups.com
> [mailto:AudioBox@yahoogroups.com] On Behalf Of jdow
> Sent: Wednesday, October 11, 2006 10:36 PM
> To: AudioBox@yahoogroups.com
> Subject: Re: [AudioBox] FW: another wild Richard idea....
> just to drive you crazy!
>
>
> If you throw enough hardware it is estimated that SoundMan
> would poop out well above a piddly eight in/eight out
> scenario. Loren tests routinely with MOTU 828 and a bank of
> three expanders.
>
> The processing speed of the attached computer is the limiting factor.
>
> {^_^}

Of course, but I only suggested limiting the number of live inputs....
not the number of outputs, because:

1.  I figured that way you wouldn't be trying to market two separate
products that are a direct competition to each other.  The AB64 would
still have the ability to route many more inputs around than SoundMan
server.  But they could both have lots of outputs.

2.  The latency involved in passing audio through the sound card to the
CPU and back again is just too large for most applications anyway.  So
unless you are just routing effects processors, or some other playback
device, etc...  I don't think live audio routing via software is ready
for prime time yet.  Even with some better sound cards that can get
really impressive latency (like RME for example) there is still about
twice as much latency (at a minimum) as most hardware digital mixers or
DSP boxes.  For some applications that is probably still good enough,
but not for any live sound, ala wireless mic routing, or orchestra mics,
etc...  depends a lot on the market I guess.

3. Routing live audio through takes a big CPU hit.  Then the lower your
latency, the more CPU you need to use. (obviously I know you and Loren
know this... I'm just giving you the reasons behind my suggestion)

4. In comparison, I wouldn't want to limit the number of outputs,
because I see it as mostly useful as a playback device.  Plus I don't
think having more outputs chews up anywhere near as much CPU as does
routing live inputs.  Unless maybe you are comparing an output with
delay and EQ, to an input with no processing.  And has always been the
case in theatre, we are usually dealing with systems that have more
outputs than inputs (at least when you are talking about playback only
shows) or at just as many outputs as inputs.

I think you get my points..   :-))

My main point however wasn't any of this.... it was to make a couple of
small changes to the Server/Audio Engine that as far as I can tell is
essentially done, and get it out the door.  :-))

Richard B. Ingraham
RBI Computers and Audio
http://www.rbicompaudio.20m.com/
rbI@...



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10/10/2006

If you throw enough hardware it is estimated that SoundMan would
poop out well above a piddly eight in/eight out scenario. Loren
tests routinely with MOTU 828 and a bank of three expanders.

The processing speed of the attached computer is the limiting
factor.

{^_^}
----- Original Message -----
From: "Richard B. Ingraham" <rbingraham@...>

>I sent this email to Charlie Richmond and Loren Wilton privatly.
> Charlie has asked me to forward it to the list.  To possiblly stimulate
> conversation.
>
> Although I've edited some things, as some of the ideas have already been
> asnwered.....
>
> -----Original Message-----
> From: Richard B. Ingraham [mailto:rbingraham@...]
>
>
> OK, so I had another random idea last night....
>
> So what if it wouldn't be too hard to make the SoundMan server, the one
> that thinks it's an Audiobox, and make it think it's an AB64 instead of
> an AB1616?  Obviously I have no idea how hard that would be, but here is
> some ideas on it:
>
> 1.  It could be made to think it's a AB64, you could simply use ABEdit
> as the UI, and you would have a product on the market that could be out
> there, and generate some interest, even if it didn't make any money yet.
>
> 2. It could actually get some use in real shows, rather than playing
> around for 30 mins at a time, or whatever the time out is right now.
>
> 3.  If the server could be made to look like a real AB64 (so it appears
> on the network, just like an real AB64 does) then you could use either
> ABShowMaker or ABEdit.  Although obviously you already need a PC to run
> the engine, so I doubt that many will want to get both a Mac and PC just
> to run one system.
>
> <snip>
>
> 5.  Maybe the SoundMan server doesn't let you have more than 8 live
> inputs, so you don't have some jackass trying to route dozens of
> channels of live audio, and asking why he gets drop outs, clicks, pops,
> etc....
>
> 6.  <snip> maybe SoundMan could become the playback only AB version?
> That's where I really see the server as the most useful anyway.
>
> 7.  This would buy Loren some time to finish his own UI, while the
> engine still gets to see the light of day.
>
> 8.  Maybe Loren just works on the engine and it's problems and someone
> else can deal with the UI that way.  (No offense Loren, I'm just
> throwing this out as a suggestion.... not because I don't think you
> could do it...)
>
>
> Ok so there it is.... delete this whenever you wish...
>
> Richard B. Ingraham
> RBI Computers and Audio
> http://www.rbicompaudio.20m.com/
> rbI@...
>
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> 10/9/2006
>
>
>
>
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> mailto:AudioBox-unsubscribe@yahoogroups.com
>
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>
> List administrator: Charlie Richmond <mailto:CharlieR@...>
> List home page: http://www.RichmondSoundDesign.com/
> Yahoo! Groups Links
>
>
>

Hi,

Thank you all for the great feedback. We'll look into the technicalities.

Cheers,

Simon







Merv Buchanan wrote:
> Charlie,
>
> This sounds like a "marriage made in Heaven" for the AB64.
>
> Good luck with this!
>
> Merv
> ----- Original Message -----
> From: "Charlie Richmond" <charlier@...>
> To: "List - AudioBox Mailing List" <AudioBox@yahoogroups.com>
> Cc: "Barry 'SFU' Truax" <truax@...>
> Sent: Tuesday, October 10, 2006 10:14 AM
> Subject: Re: [AudioBox] Using the audiobox for psychophysics research
>
>
> On Tue, 10 Oct 2006, Simon Barthelmé wrote:
>
>
>> We're currently investigating different options for running experiments
>> in audio-visual  sound localisation.  What we need is a system that
>> would allow synchronised playback of - at least - 8 sound channels with
>> microsecond precision. We're going to be manipulating phase lags between
>> the different channels, so we'd prefer the playback system not to do any
>> independent unpredictable manipulation of its own. In addition, auditory
>>
>
> The AB64 will certainly do this for up to 64 channels of playback.
>
>
>> playback will have to be synchronised with visual stimuli. We can afford
>> for audio-visual synchrony to be less precise, but an order of a few
>> milliseconds at the most is needed.
>>
>
> Individual delay adjustments are dynamically controllable in single sample
> increments.  Each sample is 2.08333333 × 10^-5 or 1/48000 second, but I'm
> sure
> you found this in the command set already, which is why you are considering
> it.
>
>
>> Sound playback would have to be controlled by a computer running Mac OS
>> X or Linux. Ideally the API would be in Matlab, but we can write our own.
>> Question : has anyone ever done anything similar using the Audiobox ? Is
>> it at all doable ?
>>
>
> I don't know how much along these lines has been done but I think it is
> certainly doable.  I suspect it's likely that anyone who has done work like
> this
> is not actually on this list because this tends to be more for exchanging
> theatrical and themed production techniques than lab experiment information.
>
> Perhaps some here might know about other such applications done in
> university
> and experimental labs, hopefully.
>
> Charlie
>
>
>> Thanks a lot,
>>
>> Simon Barthelmé
>> PhD Student
>> Laboratoire de Psychologie de la Perception
>> CNRS/Université Paris 5
>>
>
> [Non-text portions of this message have been removed]
>
>
>
> To unsubscribe from this list
> mailto:AudioBox-unsubscribe@yahoogroups.com
>
> To subscribe mailto:AudioBox-subscribe@yahoogroups.com
>
> List administrator: Charlie Richmond
> <mailto:CharlieR@...>
> List home page: http://www.RichmondSoundDesign.com/
> Yahoo! Groups Links
>
>
>
>
>
>
>
>
>
>
>
>
> To unsubscribe from this list
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>
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>
>
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>
>

I sent this email to Charlie Richmond and Loren Wilton privatly.
Charlie has asked me to forward it to the list.  To possiblly stimulate
conversation.

Although I've edited some things, as some of the ideas have already been
asnwered.....

-----Original Message-----
From: Richard B. Ingraham [mailto:rbingraham@...]
Sent: Tuesday, October 10, 2006 11:01 AM
To: 'Charlie Richmond'; 'Loren 'LASFS' Wilton'
Subject: another wild Richard idea.... just to drive you crazy!



OK, so I had another random idea last night....

So what if it wouldn't be too hard to make the SoundMan server, the one
that thinks it's an Audiobox, and make it think it's an AB64 instead of
an AB1616?  Obviously I have no idea how hard that would be, but here is
some ideas on it:

1.  It could be made to think it's a AB64, you could simply use ABEdit
as the UI, and you would have a product on the market that could be out
there, and generate some interest, even if it didn't make any money yet.

2. It could actually get some use in real shows, rather than playing
around for 30 mins at a time, or whatever the time out is right now.

3.  If the server could be made to look like a real AB64 (so it appears
on the network, just like an real AB64 does) then you could use either
ABShowMaker or ABEdit.  Although obviously you already need a PC to run
the engine, so I doubt that many will want to get both a Mac and PC just
to run one system.

<snip>

5.  Maybe the SoundMan server doesn't let you have more than 8 live
inputs, so you don't have some jackass trying to route dozens of
channels of live audio, and asking why he gets drop outs, clicks, pops,
etc....

6.  <snip> maybe SoundMan could become the playback only AB version?
That's where I really see the server as the most useful anyway.

7.  This would buy Loren some time to finish his own UI, while the
engine still gets to see the light of day.

8.  Maybe Loren just works on the engine and it's problems and someone
else can deal with the UI that way.  (No offense Loren, I'm just
throwing this out as a suggestion.... not because I don't think you
could do it...)


Ok so there it is.... delete this whenever you wish...

Richard B. Ingraham
RBI Computers and Audio
http://www.rbicompaudio.20m.com/
rbI@...

--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.408 / Virus Database: 268.13.1/469 - Release Date:
10/9/2006

Charlie,

This sounds like a "marriage made in Heaven" for the AB64.

Good luck with this!

Merv
----- Original Message -----
From: "Charlie Richmond" <charlier@...>
To: "List - AudioBox Mailing List" <AudioBox@yahoogroups.com>
Cc: "Barry 'SFU' Truax" <truax@...>
Sent: Tuesday, October 10, 2006 10:14 AM
Subject: Re: [AudioBox] Using the audiobox for psychophysics research


On Tue, 10 Oct 2006, Simon Barthelmé wrote:

> We're currently investigating different options for running experiments
> in audio-visual  sound localisation.  What we need is a system that
> would allow synchronised playback of - at least - 8 sound channels with
> microsecond precision. We're going to be manipulating phase lags between
> the different channels, so we'd prefer the playback system not to do any
> independent unpredictable manipulation of its own. In addition, auditory

The AB64 will certainly do this for up to 64 channels of playback.

> playback will have to be synchronised with visual stimuli. We can afford
> for audio-visual synchrony to be less precise, but an order of a few
> milliseconds at the most is needed.

Individual delay adjustments are dynamically controllable in single sample
increments.  Each sample is 2.08333333 × 10^-5 or 1/48000 second, but I'm
sure
you found this in the command set already, which is why you are considering
it.

> Sound playback would have to be controlled by a computer running Mac OS
> X or Linux. Ideally the API would be in Matlab, but we can write our own.
> Question : has anyone ever done anything similar using the Audiobox ? Is
> it at all doable ?

I don't know how much along these lines has been done but I think it is
certainly doable.  I suspect it's likely that anyone who has done work like
this
is not actually on this list because this tends to be more for exchanging
theatrical and themed production techniques than lab experiment information.

Perhaps some here might know about other such applications done in
university
and experimental labs, hopefully.

Charlie

>
> Thanks a lot,
>
> Simon Barthelmé
> PhD Student
> Laboratoire de Psychologie de la Perception
> CNRS/Université Paris 5

[Non-text portions of this message have been removed]



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<mailto:CharlieR@...>
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It depends on how you define "similar".  The basic concept of syncronized
tracks (which are locked on a sample basis, considerably better than
millisecond accuracy) and synchronizing them to extrrnal things is what
theatrical audio reproduction is all about.  The AB can definitely do what
you want.

There almost certainly isn't a Matlab interface currently, but as you
mention, one can be written fairly easily.

         Loren

On Tue, 10 Oct 2006, Simon Barthelmé wrote:

> We're currently investigating different options for running experiments
> in audio-visual  sound localisation.  What we need is a system that
> would allow synchronised playback of - at least - 8 sound channels with
> microsecond precision. We're going to be manipulating phase lags between
> the different channels, so we'd prefer the playback system not to do any
> independent unpredictable manipulation of its own. In addition, auditory

The AB64 will certainly do this for up to 64 channels of playback.

> playback will have to be synchronised with visual stimuli. We can afford
> for audio-visual synchrony to be less precise, but an order of a few
> milliseconds at the most is needed.

Individual delay adjustments are dynamically controllable in single sample
increments.  Each sample is 2.08333333 × 10^-5 or 1/48000 second, but I'm sure
you found this in the command set already, which is why you are considering it.

> Sound playback would have to be controlled by a computer running Mac OS
> X or Linux. Ideally the API would be in Matlab, but we can write our own.
> Question : has anyone ever done anything similar using the Audiobox ? Is
> it at all doable ?

I don't know how much along these lines has been done but I think it is
certainly doable.  I suspect it's likely that anyone who has done work like this
is not actually on this list because this tends to be more for exchanging
theatrical and themed production techniques than lab experiment information.

Perhaps some here might know about other such applications done in university
and experimental labs, hopefully.

Charlie

>
> Thanks a lot,
>
> Simon Barthelmé
> PhD Student
> Laboratoire de Psychologie de la Perception
> CNRS/Université Paris 5

[Non-text portions of this message have been removed]

Hi,

We're currently investigating different options for running experiments
in audio-visual  sound localisation.  What we need is a system that
would allow synchronised playback of - at least - 8 sound channels with
microsecond precision. We're going to be manipulating phase lags between
the different channels, so we'd prefer the playback system not to do any
independent unpredictable manipulation of its own. In addition, auditory
playback will have to be synchronised with visual stimuli. We can afford
for audio-visual synchrony to be less precise, but an order of a few
milliseconds at the most is needed.
Sound playback would have to be controlled by a computer running Mac OS
X or Linux. Ideally the API would be in Matlab, but we can write our own.
Question : has anyone ever done anything similar using the Audiobox ? Is
it at all doable ?

Thanks a lot,

Simon Barthelmé
PhD Student
Laboratoire de Psychologie de la Perception
CNRS/Université Paris 5

On Wed, 21 Jun 2006, Richard B. Ingraham wrote:

> Fostex already has some units that do that.  If they are actually are
> shipping that stuff.
>
> http://www.netcira.com/docs/home/netcira_front.shtml
>
> There are some of the things I've been looking at, if I was going down
> that path.
>
> About how much do you think the AB64 ES I/O cards will run?

The only thing I know for sure is they are somewhat cheaper than the Cobranet
ones.

Charlie

Fostex already has some units that do that.  If they are actually are
shipping that stuff.

http://www.netcira.com/docs/home/netcira_front.shtml

There are some of the things I've been looking at, if I was going down
that path.

About how much do you think the AB64 ES I/O cards will run?

Richard B. Ingraham
RBI Computers and Audio
http://www.rbicompaudio.20m.com
rbi@...

> -----Original Message-----
> From: AudioBox@yahoogroups.com
> [mailto:AudioBox@yahoogroups.com] On Behalf Of Charlie Richmond
> Sent: Wednesday, June 21, 2006 11:49 AM
> To: List - AudioBox Mailing List
> Subject: [AudioBox] digital interfaces
>
>
> Hi folks -
>
> I had a long chat with Jimmy Kawalek at EtherSound and they
> look like the best
> bet for coming up with various digital interfaces to the
> AB64.  They currently
> have a variety of AES/EBU boxes and are seriously considering
> MADI since they
> are getting a fair number of requests for that.  I put in a
> request for an ES to
> ADAT and they will discuss this today but say that they have
> never been asked
> for this before!
>
> In the meantime, we expect the first shipment of ESX cards to
> be tested in
> Vancouver 'real soon now'...
>
> The range of ES products currently available are listed here:
>
http://www.ethersound.com/products/product.php

And this seems to be the way to go now.

Charlie



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Yahoo! Groups Links

Hi folks -

I had a long chat with Jimmy Kawalek at EtherSound and they look like the best
bet for coming up with various digital interfaces to the AB64.  They currently
have a variety of AES/EBU boxes and are seriously considering MADI since they
are getting a fair number of requests for that.  I put in a request for an ES to
ADAT and they will discuss this today but say that they have never been asked
for this before!

In the meantime, we expect the first shipment of ESX cards to be tested in
Vancouver 'real soon now'...

The range of ES products currently available are listed here:

http://www.ethersound.com/products/product.php

And this seems to be the way to go now.

Charlie

On Tue, 13 Jun 2006, Charlie Richmond wrote:

> I have been given a quote for an ADAT card that can plug in to the digital
> I/O slot of the AB64.  It's not cheap but could well be worth it.  Each card
> would have 32 In and 32 Out (in other words 4 separate ADAT connections) and
> all 64 in and out channels of a fully loaded AB64 can be handled with two of
> these cards.

Based on the responses from the first post, we have decided to design this as a
two part unit - a single card that plugs into one of the digital I/O slots and a
breakout box with 8 connections on it so that all 64 I/Os are on a single I/F.
This would cost twice as much as the original unit but only take up one slot so
the other can be used for a Cobranet or Ethersound interface simultaneously.

Because people seemed to think the cost was not a factor, this is how we will
go.  Thanks for you input, everyone!


Charlie

---------- Forwarded message ----------
Date: Fri, 16 Jun 2006 23:30:00 +0200
From: Martin Lukesch
To: Charlie Richmond <charlier@...>
Subject: MADI

Charlie,


I use MADI, and every year more. So increase. My complete recording
equipment is linked via MADI. One MADI fibre goes from the NEXUS matrix
(and mic preamps) to a RME MADI Brige, the Sony mixer (with MADI card)
is connected to the Bridge, 2 DAWs with RME MADIPCI cards (and Nuendo
software) are connected to the MADI Bridge, one Alesis HD2 is connectes
via a RME MADI to Adat Converter to the MADI Bridge. Works pretty well,
we do all our recordings things with this equipment.

Not the best solution, but better than anything else in the moment. I am
dreaming of the self configuring audio + data connection, but maybe this
works in our next life.

bye

On Thu, 15 Jun 2006, Richard B. Ingraham wrote:

>> There's also the issue that ADAT mic pres are 'consumer
>> grade' usually and
>> ES/Cobra are 'professional' so you should get more quality in
>> that area too.  Of
>> course this may not be an issue for most live shows.
>>
>
> I would disagree with that fairly highly.  There are all kinds of
> "grades" of ADAT mic pres.  Some I'm sure are crap.  Some can cost your

OK, thanks for informing me.  The above is simply what I was told a while ago.

>>  How many amps
>> and/or powered speakers have ADAT inputs?????  none, I'd
>> wager, so this is not
>> going to make any difference to our main market.
>>
> I've never seen an amp or a powered speaker with ADAT inputs.  I can see
> your point.

And this is the biggest problem I have with investing in this right now.  We
just need more people seriously saying they can use ADAT connections right at
the moment and not that they 'take them or leave' them and then provide more
arguments like you have as to why analog is the still the best common
denominator.

Thanks as always!
Charlie

> -----Original Message-----
> From: AudioBox@yahoogroups.com
> [mailto:AudioBox@yahoogroups.com] On Behalf Of Charlie Richmond
> Sent: Thursday, June 08, 2006 5:01 AM
> To: AudioBox@yahoogroups.com
> Subject: RE: [AudioBox] digital interfaces...
>

> There's also the issue that ADAT mic pres are 'consumer
> grade' usually and
> ES/Cobra are 'professional' so you should get more quality in
> that area too.  Of
> course this may not be an issue for most live shows.
>

I would disagree with that fairly highly.  There are all kinds of
"grades" of ADAT mic pres.  Some I'm sure are crap.  Some can cost your
first born child.  Some I know are being used by the folks that record
the Cleveland Orchestra concerts, (although modified to use glass fiber
optic rather than the cheap plastic)and I've met those guys..... they
have a fairly discerning ear....  :-)

As for ES/Cobra, there are some cheap mic pres with those as well.
Whirlwind specifically told us that their one product that has something
like a pair of mic preamps onto a Cobra system, was for use in paging
and the like only.  It was not good enough for sound reinforcement.

Although most of those boxes cost a lot more than the ADAT units, I
seriously doubt the pres themselves are of any higher quality in
general.  In fact I would have more faith in buying a Presonus mic pre
any day over something made by Digigram.  If I was buying just on name
alone, and not actually getting my hands on a product.


> This might be doable.  I have asked.  The weird thing is that
> the most common
> thing most people want to connect to AB outputs is
> loudspeakers and there are
> very few Cobra/ES self powered loudspeakers, which would seem
> to be the most
> logical way to go.  There are also not many low cost power
> amps with that input
> option either so people still have to go with analog outputs.

This I completely agree with.  This is why I frankly think that it all
sort of "comes out in the wash" so to speak.  In other words, you have
to pay for that analog conversion at some point in the chain.  And while
I can save quite a bit of money by getting an AB64 with one of the
digital I/O options, rather than just analog I/O, I have to spend so
much more on other components that can talk to these various digital
I/Os, that I end up saving nothing, or in some cases spending more.

That is why I think all this Cobra and ES stuff only serves well for
those folks like myself that would really enjoy having the digital
interconnectivity, and think the benefits are worth it, to only
converting from A to D and D to A once in the entire sound system.  And
they are also willing to spend a bit more to get that.

Or the other instance is where you really can save a bundle on
installation costs, such as convention centers, arenas, theme parks and
the like, where you have to install miles of cables, and the less you
have to pull the better.

This is why for the upcoming projects I am working on, my current
thoughts would be to go with both analog and digital I/O for the AB64.
That way I could have the flexibility of routing stuff from one or more
digital consoles to the AB64.  The AB64 would then act like the system D
to A unit, and plug directly into amps in the same audio rack, or it
would send signals to powered speakers.

I know I could hang some small nodes for digital outputs near the
various speaker locations and get a digital feed until really close to
the speakers.  But that seems a bit of a hassle to me, and I'm not sure
it's worth it frankly.  Unless I could find some units that also allowed
power over the Cat 5 cable, and I know some of them do that.  So that
way you don't necessarily need a power outlet where you have a node that
takes signals off the audio network, and converts to analog.

But really....  I'm just not all that big of a fan of powered
loudspeakers in general.  Too many times I've worked in a space where
powered speakers were used, and I almost always have wished at some
point that I could use that power amp channel for something else.  Or I
wish I could gang the two self powered speakers together on a single
channel and use the other channel for a special speaker some place.
Which of course you could easily do, if your using a conventional amp
and speaker set-up.  But no way to do that, if all or most of your
speakers are self powered.




>  How many amps
> and/or powered speakers have ADAT inputs?????  none, I'd
> wager, so this is not
> going to make any difference to our main market.
>
I've never seen an amp or a powered speaker with ADAT inputs.  I can see
your point.

Richard B. Ingraham
RBI Computers and Audio
http://www.rbicompaudio.20m.com
rbi@...

YMMV indeed.

As I said, when it works, it's fine, but when it doesn't it's not fun at
all, which is why I think that there needs to be some serious testing.
There's one particular set up that gives me grief constantly and that's an
Akai S6000 to Yamaha 02R. Totally random changes of level, word clock
problems, etc. Other set-ups are solid, as you've experienced.

Regards,

--
John Leonard
Sound & Show Control
10 Belsize Park
Hampstead
London
NW3 4ES
United Kingdom


T: +44 (0)20 7794 5942
F: +44 (0)20 7431 4716
M: +44 (0)7774 758774
Skype:  soundmanjohn
SkypeIn: +44 (0)20 8816 7587
IATSE Local 1 Card#00569



> From: "Richard B. Ingraham" <rbingraham@...>
> Reply-To: <AudioBox@yahoogroups.com>
> Date: Wed, 14 Jun 2006 00:57:02 -0400
> To: <AudioBox@yahoogroups.com>
> Subject: RE: [AudioBox] ADAT card ...
>
> I guess it just proves that YMMV?  :-)

I would think that if you could actually afford a 32 or 64 channel
Audiobox 64 set up, the cost of the digital I/O card is pretty
insignificant.  And for those of us that might want it, I would guess
that they are not buying the digital I/O option, just to save some bucks
over the analog interface.  I think it would simply be because they want
to integrate it into some kind (whether that be Cobra, or ES, or ADAT,
or whatever) digital audio network, or digital audio connection system,
just so they don't have to convert to analog and back again, like you
did with earlier ABs.  I doubt it's because they look at the price list,
and say.... whoa...  I can save a bundle if I go with ES or ADAT over
standard analog I/O.  In other words I think it's more of a convenience
thing,  or "cool" thing, than a $$ saving thing.

If you have $14270 to spend for a 64 channel box.  I would bet that the
price difference of $8776 between 64 channels of analog and 64 channels
of Cobra (just to use an example, since ES nor the ADAT is on the web
site price list) is not going to be that much of a sticker shock, or a
deal breaker.  But more like... huh...  I want to go digital... and oh
look, I can save some money with that method as well.... OK that solves
it, I'll go digital.    I doubt it goes like.... well I really just want
64 channels of analog I/O, but I'm going to go digital since I can save
$8K.  Especially since you have to have the cost of the A-D-A somewhere,
and if you don't have it in the AB64, it will just be some place else in
the signal chain.  (assuming that the buyer understands that
concept/reality)  Although I guess there would be cost savings over have
A-D-A at multiple places within a sound system, that are needed if a
digital I/O option is available.  ;-)

I'm thinking this based on my understanding of a the major markets for
the AB64.  Poor theatres just scraping by won't be picking themselves up
even a 16 channel AB64 any time soon.  ;-)

At least that's my  opinion.

Richard B. Ingraham
RBI Computers and Audio
http://www.rbicompaudio.20m.com
rbi@...

> -----Original Message-----
> From: AudioBox@yahoogroups.com
> [mailto:AudioBox@yahoogroups.com] On Behalf Of Merv Buchanan
> Sent: Tuesday, June 13, 2006 4:02 AM
> To: AudioBox@yahoogroups.com
> Subject: Re: [AudioBox] ADAT card ...
>
>
> Well, let's put it out there and see what happens. To give us
> some wiggling room we could call it "preliminary pricing".
> ----- Original Message -----
> From: "Charlie Richmond" <charlier@...>
> To: <AudioBox@yahoogroups.com>
> Sent: Tuesday, June 13, 2006 3:37 PM
> Subject: Re: [AudioBox] ADAT card ...
>
>
> > On Tue, 13 Jun 2006, Merv Buchanan wrote:
> >
> > > Can we manage pricing similar to our other cards, or is
> the cost too
> hign?
> >
> > It looks like we will have to do that and hopefully others will feel
> that's
> > appropriate.
> >
> > Charlie
> >
> >
> >
> > To unsubscribe from this list
> > mailto:AudioBox-unsubscribe@yahoogroups.com
> >
> > To subscribe mailto:AudioBox-subscribe@yahoogroups.com
> >
> > List administrator: Charlie Richmond
> <mailto:CharlieR@...>
> > List home page: http://www.RichmondSoundDesign.com/
> > Yahoo! Groups Links
> >
> >
> >
> >
> >
> >
> >
> >
>
>
>
>
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