RE: [linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-31 Thread Richard C. Burnett

Thought I would chime in just for some extra information in case anyone
cared to know.  As the feature size of chips keeps getting smaller and
smaller, these same issues are getting increasingly worse!  At 0.18 signal
integrity doesn't always have to be checked if you are careful with your
rise and fall times for digital signal (plus your fanout), but as you go
to 0.13um feature size, the capacitive effects start to cause big signal
integrity problems.  It is interesting to me that these problems are a big
problem at the board level too, I didn't know that.

Rick



On Thu, 30 May 2002, Bob Colwell wrote:

 Somehow I feel the need to get even more specific.
 
 Voltage sags for two reasons: DC resistance and AC impedance.
 DC resistance is just plain old ohms law -- no perfect conductors
 exist (until the folks in the labs give us high-temp superconductors),
 so a current flow will always cause a voltage drop, as in E=IR.
 
 AC impedance matters because conductors also exhibit inductance and
 capacitance. Capacitance causes signal leakage and cross-coupling.
 But inductance causes AC-induced voltage spikes, because coils
 convert electrical energy into a magnetic field, and if you try to
 collapse that magnetic field quickly (as a fast-changing electrical
 signal does) it reconverts back into an electrical potential that
 acts to oppose the change in current. If one distributes capacitors
 of the appropriate size and speed, one can effectively decouple
 the downstream inductances from the current spike.
 
 Lamar knows all this but I thought it might benefit others who might
 not. -BobC
 

++---+
|  T a l i t y   |  +--+ |
++ ++-+| |
| Richard Burnett  |+-+| |
|  Senior Design Engineer  +---+  ++ |
|   [EMAIL PROTECTED] |  |  |
|  |  |  |
| Phone: 919.380.3014 |  |
|   Fax: 919.380.3903  |  |  |
++




RE: [linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-30 Thread Bob Colwell

Somehow I feel the need to get even more specific.

Voltage sags for two reasons: DC resistance and AC impedance.
DC resistance is just plain old ohms law -- no perfect conductors
exist (until the folks in the labs give us high-temp superconductors),
so a current flow will always cause a voltage drop, as in E=IR.

AC impedance matters because conductors also exhibit inductance and
capacitance. Capacitance causes signal leakage and cross-coupling.
But inductance causes AC-induced voltage spikes, because coils
convert electrical energy into a magnetic field, and if you try to
collapse that magnetic field quickly (as a fast-changing electrical
signal does) it reconverts back into an electrical potential that
acts to oppose the change in current. If one distributes capacitors
of the appropriate size and speed, one can effectively decouple
the downstream inductances from the current spike.

Lamar knows all this but I thought it might benefit others who might
not. -BobC

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Lamar Owen
Sent: Tuesday, May 28, 2002 4:49 PM
To: [EMAIL PROTECTED]
Subject: Re: [linux-audio-dev] Writing a driver for this card: your
thoughts?


On Tuesday 28 May 2002 07:20 pm, Tobias Ulbricht wrote:
 one question:
 could you do this for the unsophisticated audiophile with less Acronyms?
   supply. Using a computer psu is going to severely limit what you are
  ^^^ = power supply unit?

Yes.

  decoupling is done at the amplifier rails, along with 'sag' compensating
  caps
  ^^^

Voltage sag due to current draw.

  (with low ESR! Tantalum only, and preferably sintered slug) in the
proper
 ^^^ ? electron spin resonance :)?

Equivalent series resistance.  The higher the ESR, the slower the cap will
discharge, and the less effective it is as a 'voltage flywheel'.

  But again, the hardware must be able to cancel out the enevitable RFI
  that is
 ^^^

Radio Frequency Interference.

Sorry.
--
Lamar Owen
WGCR Internet Radio
1 Peter 4:11




Re: [linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-29 Thread Paul Winkler

On Tue, May 28, 2002 at 07:43:14PM -0400, Lamar Owen wrote:
 Class A design has it's strong points.  As long as you are in a good linear 
 portion of the transconductance curve you're OK.

As a sometime guitarist, I'd say Class A has its strong points well
into distortion...
Hey Taybin, I know you weren't really into that Sleater-Kinney show,
but how did you like that Vox AC-30 that Carrie plays through?
Yum!

-- 

Paul Winkler
home:  http://www.slinkp.com
Muppet Labs, where the future is made - today!



Re: [linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-29 Thread Taybin Rutkin

On Wed, 29 May 2002, Paul Winkler wrote:

 Hey Taybin, I know you weren't really into that Sleater-Kinney show,
 but how did you like that Vox AC-30 that Carrie plays through?

Oh yeah!  Her sound was great.  Real warm.

Taybin




Re: [linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-29 Thread Lamar Owen

On Wednesday 29 May 2002 11:18 am, Paul Winkler wrote:
 On Tue, May 28, 2002 at 07:43:14PM -0400, Lamar Owen wrote:
  Class A design has it's strong points.  As long as you are in a good
  linear portion of the transconductance curve you're OK.

 As a sometime guitarist, I'd say Class A has its strong points well
 into distortion...

But then the amplifier is *producing* sound rather than *reproducing* sound, 
and has become a form of compressor.  A compressor with some interesting 
curves, maybe, but a compressor nonethless.  And a compressor one can model 
and produce identical characteristics in effects plugins. :-)

As a sometimes guitar player myself, that is.  Still have my copy of 
Anderton's 'Electronic Projects for Musicians' lying around here.
-- 
Lamar Owen
WGCR Internet Radio
1 Peter 4:11



Re: [linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-29 Thread Paul Winkler

On Wed, May 29, 2002 at 02:22:09PM -0400, Lamar Owen wrote:
 On Wednesday 29 May 2002 11:18 am, Paul Winkler wrote:
  On Tue, May 28, 2002 at 07:43:14PM -0400, Lamar Owen wrote:
   Class A design has it's strong points.  As long as you are in a good
   linear portion of the transconductance curve you're OK.
 
  As a sometime guitarist, I'd say Class A has its strong points well
  into distortion...
 
 But then the amplifier is *producing* sound rather than *reproducing* sound, 
 and has become a form of compressor.  A compressor with some interesting 
 curves, maybe, but a compressor nonethless.  And a compressor one can model 
 and produce identical characteristics in effects plugins. :-)

In theory, yes. In practice, I think we're still in the close but no
cigar stage of modelling amplifiers. But I do think we'll get it
eventually.

--

Paul Winkler
home:  http://www.slinkp.com
Muppet Labs, where the future is made - today!



Re: [linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-29 Thread Lamar Owen

On Wednesday 29 May 2002 02:50 pm, Paul Winkler wrote:
 On Wed, May 29, 2002 at 02:22:09PM -0400, Lamar Owen wrote:
  interesting curves, maybe, but a compressor nonethless.  And a compressor
  one can model and produce identical characteristics in effects plugins.
  :-)

 In theory, yes. In practice, I think we're still in the close but no
 cigar stage of modelling amplifiers. But I do think we'll get it
 eventually.

If one has the amplifier in question, one can produce a reasonable facsimile 
of the transfer function using various waveforms and frequencies, then 
interpolate (cubic splines, probably) to determine a transfer function that 
one cane then put into a 'transfer function engine' that would replicate said 
transfer function.  You should be able to characterize down to the individual 
lot number level with enough resolution in the interpolation.  

Has such a 'transfer function engine' effect plugin been written?
-- 
Lamar Owen
WGCR Internet Radio
1 Peter 4:11



Re: [linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-29 Thread Fred Gleason

On Wednesday 29 May 2002 14:22, Lamar Owen wrote:

 But then the amplifier is *producing* sound rather than *reproducing*
 sound, and has become a form of compressor.  A compressor with some
 interesting curves, maybe, but a compressor nonethless.  And a compressor
 one can model and produce identical characteristics in effects plugins. :-)

Fie fie, you infidel!  :)

Cheers!


|-|
|Frederick F. Gleason, Jr.|WAVA Radio - 105 FM |Voice: 1-(703)-807-2266   |
| Director of Engineering |1901 N. Moore Street|  FAX: 1-(703)-807-2245   |
| |Arlington, VA 22209 |  Web: HTTP://www.wava.com|
|-|
|DALLAS:  |
|   The city that chose Astroturf to  |
|   keep the cheerleaders from grazing.   |
|-|



Re: [linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-28 Thread Tim Orford

On Mon, May 27, 2002 at 10:48:28PM -0400, Lamar Owen wrote:
 Properly designed balanced in and out A/D and D/A converters can be completely 
 immune to the PC's internal noise.  But it all goes to proper design.  The 
 Antex SX series of cards likewise contain internal A/D converters.  I have 
 measured the noise figures of the Antex SX-36 we have at WGCR using state of 
 the art audio systems analyzers, and the noise floor is below the LSB 
 threshold.  All due to balanced I/O, sound PC layout techniques, and top of 
 the line components.

Just to try and put things in perspective, surely once you go beyond a certain
level, signal-unrelated noise is one of the least important indications of 
sound quality? For me anyway. The quality of the noise is important
also :-)


 In a high RF environment, unless the converters are optically isolated from 
 the PC, you might be asking for trouble.  When I say 'high RF' I'm talking 
 10KW of AM transmitter fifteen feet away, with a measured RF field intensity 
 of 105V/m (the ANSI exposure limit is around 645V/m).  This means a one meter 
 piece of wire that isn't properly grounded can develop 105V of RF energy.  I 
 have suffered RF burns of appreciable intensity touching wires that weren't 
 connected to anything on either end -- they were just oriented along the 
 field gradient.

your requirements do seem to be rather different than a recording studio
:-)


 Also, due to the processing typical radio stations use in the air chain, in 
 many cases the quality of playback and record hardware for on-air use must be 
 up to the top quality of a recording studio.  Typical radio stations use 
 sophisticated compressors, multiband limiters, and many other DSP-driven 
 techniques to maximize loudness -- and when the dynamics of the source 
 material go pianissimo, the compressors drive up the noise floor to 
 compensate.  Noise in the outputs of a broadcast automation or 
 music-on-hard-drive system is verboten. []

such insane amounts of compression as used by most radio stations are not
usually related to 'quality' as understood by most poeple.

one of the most important aspects of analogue design is a good power
supply. Using a computer psu is going to severely limit what you are
going to achieve. Regulators are by no means perfect.

also, perhaps academic here, but using balanced
audio can actually degrade the sound by cancelling out even order
harmonics but leaving the odd order ones, or by increasing the component
count. Very high end stuff designed for non-hostile environments tends
to be unbalanced.

sorry if i'm just being pedantic :-)


cheers

Tim Orford


ok, back to trying to get Ardour compiled.(does that make me
On-Topic now?)




Re: [linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-28 Thread Fred Gleason

On Friday 24 May 2002 22:11, Paul Davis wrote:

 By my standards, anybody who puts A-D/D-A conversion inside their
 computer chassis is by definition not concerned with the best in the
 business. The term professional audio for me doesn't include analog
 I/O for a computer card unless the converters are in a separate
 box. It might be good enough for a radio station, but for a recording
 studio?

Sounds like you've never heard one of these cards in action.  The entire 
analog section is mu-metal shielded, and in terms of quality can stand with 
the best of the external designs.  Personally, I define professional as 
that which meets a certain standard of quality in operation and features, 
which these cards certainly do.  *How* that is achieved is really irrelevant.

BTW, these cards work very well under Linux right now, using AudioScience's 
HPI interface.

Cheers!


|-|
|Frederick F. Gleason, Jr.|WAVA Radio - 105 FM |Voice: 1-(703)-807-2266   |
| Director of Engineering |1901 N. Moore Street|  FAX: 1-(703)-807-2245   |
| |Arlington, VA 22209 |  Web: HTTP://www.wava.com|
|-|
| Some changes are so slow, you don't notice them.|
| Others are so fast, they don't notice you.  |
|  -- Anonymous   |
|-|



Re: [linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-28 Thread Lamar Owen

[I may regret this... :-)]

On Tuesday 28 May 2002 01:19 pm, Tim Orford wrote:
 Lamar Owen wrote:
  Also, due to the processing typical radio stations use in the air chain,
  in many cases the quality of playback and record hardware for on-air use
  must be up to the top quality of a recording studio.

 such insane amounts of compression as used by most radio stations are not
 usually related to 'quality' as understood by most poeple.

What happens is that lower quality audio is made to sound that much worse by 
that very compression.  Reverb is exaggerated, etc.  As to the amounts being 
insane -- well, that is a big point of contention in the broadcast engineer 
and broadcast program director communities.  

 one of the most important aspects of analogue design is a good power
 supply. Using a computer psu is going to severely limit what you are
 going to achieve. Regulators are by no means perfect.

If the output impedance of the output stage is low enough, and proper 
decoupling is done at the amplifier rails, along with 'sag' compensating caps 
(with low ESR! Tantalum only, and preferably sintered slug) in the proper 
places, DC rail regulation becomes moot.  The key is a properly designed 
feedback network, with low net gain, using a high quality op amp with as high 
an open loop gain as possible, and operating the output well below rail 
voltage.  If the output voltage goes over half rail, then all bets are off.

 also, perhaps academic here, but using balanced
 audio can actually degrade the sound by cancelling out even order
 harmonics but leaving the odd order ones, or by increasing the component
 count. Very high end stuff designed for non-hostile environments tends
 to be unbalanced.

Phase-linear balanced outputs won't cancel _any_ harmonics.  Where did that 
concept come from?  And the Carver Silver Seven sounds better than the old 
M400, too, right? :-) (audiophile humor)

 ok, back to trying to get Ardour compiled.(does that make me
 On-Topic now?)

Hmmm.  You know, a linux based broadcast compressor (on the order of an Omnia6 
or Optimod 8400) would be an interesting project.  Someone who wants to do 
serious DSP work could have a field day doing multiband limiting and the 
various other things broadcast processors must do.

But again, the hardware must be able to cancel out the enevitable RFI that is 
going to be present.
-- 
Lamar Owen
WGCR Internet Radio
1 Peter 4:11



Re: [linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-28 Thread Fred Gleason

On Tuesday 28 May 2002 16:19, Lamar Owen wrote:

 Hmmm.  You know, a linux based broadcast compressor (on the order of an
 Omnia6 or Optimod 8400) would be an interesting project.  Someone who wants
 to do serious DSP work could have a field day doing multiband limiting and
 the various other things broadcast processors must do.

Actually, I just heard an interesting rumor.  One of the major broadcast 
transmitter manufacturers (the one with sales offices in Mason OH) is 
implementing their exciter for the new Ibiquity FM DAB system on an embedded 
Linux platform.  This includes some very high end encoding technologies (like 
PAC/AAC).  It'll be interesting indeed to see how this pans out.

Cheers!


|-|
|Frederick F. Gleason, Jr.|WAVA Radio - 105 FM |Voice: 1-(703)-807-2266   |
| Director of Engineering |1901 N. Moore Street|  FAX: 1-(703)-807-2245   |
| |Arlington, VA 22209 |  Web: HTTP://www.wava.com|
|-|
|The real problem with hunting elephants is carrying the decoys.  |
| -- Anonymous|
|-|



Re: [linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-28 Thread Tobias Ulbricht


one question:
could you do this for the unsophisticated audiophile with less Acronyms?
What's all that:

  supply. Using a computer psu is going to severely limit what you are
 ^^^ = power supply unit?

 decoupling is done at the amplifier rails, along with 'sag' compensating caps
 ^^^

 (with low ESR! Tantalum only, and preferably sintered slug) in the proper
^^^ ? electron spin resonance :)?

 But again, the hardware must be able to cancel out the enevitable RFI that is
^^^
thanks, tobias.




Re: [linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-28 Thread Lamar Owen

On Tuesday 28 May 2002 06:40 pm, Tim Orford wrote:
 On Tue, May 28, 2002 at 04:19:48PM -0400, Lamar Owen wrote:
 :-) bring on the flames :-)

This is a flame?  Tempest in a teacup... :-)

 and good quality audio sounds better?
 this is a new one to me - i've never heard anyone
 blame it on the source material beforehmmm...:-)

Yes, actually, good quality source material sounds better.  Orban's Optimod 
9100 manual, circa 1987, mentions this as part of the setup instructions.

 I do sympathise with you, but if that much compression was
 desirable, then the tracks would be mixed like that in the first place.

Have you checked the dynamic range (or lack thereof) on some recent CD's?  
They are compressed flat as a pancake.

But, like I said, for AM radio coverage area is directly proportional to 
modulation percentage.  FM radio doesn't have that excuse.

And I personally process the least I can get away with.  But it does 
compensate for much operator error in the lack of watching levels.  And the 
six band limiter of the Optimod 9100 is pretty clean.

 But like you said, it has been a big point of contention for a long
 time, and i dont think we will solve it here :-)

Yes, indeed.

  Phase-linear balanced outputs won't cancel _any_ harmonics.  Where did
  that concept come from?

 i have seen this in balanced internal circuitry, for example
 in valve compressors. Its been a few years since i did much
 analogue stuff, so i'm gonna be deliberately vague here.

Certainly in a non-linear situation (such as a tube compressor, such as the 
ever popular UREI leveling amplifier (got one)) balanced versus non-balanced 
internal to the processing will change the transfer function.  But this is an 
I/O discussion.

 And the Carver Silver Seven sounds better than the old
  M400, too, right? :-) (audiophile humor)

 ok, i can see that you think that i am some kind of audiophile
 lunatic, which maybe i am to a certain extent. There have been
 several things which have changed my life in a major way in the
 last few years:

That wasn't my intention; my apologies.

 One was the discovery of the Free Software movement. One was the
 discovery of class A, low feedback design (thereby shattering
 most of what i was taught at college).

Class A design has it's strong points.  As long as you are in a good linear 
portion of the transconductance curve you're OK.

 I'm not really disagreeing with you, but i think my
 point was that there are more factors to take into
 account that you appeared to be ignoring. 

There's always factors.  In another forum I might detail how the assymetry of 
tower base impedance creates distortion in the transmitted audio of an AM 
station, and how silver plated coils and multi-thousand dollar vacuum 
capacitors with computer modeled matching networks have more of a bearing on 
my sound than the audio inside the studio.  My matching network here is worth 
over $75,000, thanks to the engineering behind it.  And that's just so that 
my 11khz sliver of audio passband will be as flat as possible. But that would 
be a major digression.  By private e-mail, perhaps.

There is a lot
 more to analogue audio than just keeping impedances low,
 wacking up the negative feedback, and measuring the noise
 floor, as i'm sure you realise.

Certainly.  Transmission line factors must be taken into account for long runs 
at high frequencies, etc.  Lots of factors.  But the question I was answering 
was a simple 'I've never heard of balanced drivers; what are they?'

 So to get back to the original point, i'm sure the cards
 you mentioned are indeed fine cards, but if cost is no object
 (ha ha!) and if quality is your objective, then you will put your
 analogue converters in a separate box with a dedicated power
 supply.

Those ASI cards ain't cheap.  Check them out: audioscience.com.
-- 
Lamar Owen
WGCR Internet Radio
1 Peter 4:11



Re: [linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-28 Thread Lamar Owen

On Tuesday 28 May 2002 07:20 pm, Tobias Ulbricht wrote:
 one question:
 could you do this for the unsophisticated audiophile with less Acronyms?
   supply. Using a computer psu is going to severely limit what you are
  ^^^ = power supply unit?

Yes.

  decoupling is done at the amplifier rails, along with 'sag' compensating
  caps
  ^^^

Voltage sag due to current draw.

  (with low ESR! Tantalum only, and preferably sintered slug) in the proper
 ^^^ ? electron spin resonance :)?

Equivalent series resistance.  The higher the ESR, the slower the cap will 
discharge, and the less effective it is as a 'voltage flywheel'.

  But again, the hardware must be able to cancel out the enevitable RFI
  that is
 ^^^

Radio Frequency Interference.

Sorry.
-- 
Lamar Owen
WGCR Internet Radio
1 Peter 4:11



Re: [linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-27 Thread Lamar Owen

On Friday 24 May 2002 10:11 pm, Paul Davis wrote:
 As to the AudioScience cards, they are common in radio stations and
  recording studios.

 By my standards, anybody who puts A-D/D-A conversion inside their
 computer chassis is by definition not concerned with the best in the
 business. The term professional audio for me doesn't include analog
 I/O for a computer card unless the converters are in a separate
 box. It might be good enough for a radio station, but for a recording
 studio?

Properly designed balanced in and out A/D and D/A converters can be completely 
immune to the PC's internal noise.  But it all goes to proper design.  The 
Antex SX series of cards likewise contain internal A/D converters.  I have 
measured the noise figures of the Antex SX-36 we have at WGCR using state of 
the art audio systems analyzers, and the noise floor is below the LSB 
threshold.  All due to balanced I/O, sound PC layout techniques, and top of 
the line components.

With unbalanced I/O all bets are off, of course.

But I still use an Echo Layla in our production studio -- and Layla's A/D is 
out of the PC in a rackmount chassis.  Balanced I/O on TRS 1/4 inch jacks.  
But no Linux drivers yet.

 I think you're always better off with 3rd party converters myself, so
 you can make you're own choices on the quality level, and possibly
 have multiple options (Fostex,Frontier Designs,Apogee for example) as
 well as upgrade/downgrade paths.

In a high RF environment, unless the converters are optically isolated from 
the PC, you might be asking for trouble.  When I say 'high RF' I'm talking 
10KW of AM transmitter fifteen feet away, with a measured RF field intensity 
of 105V/m (the ANSI exposure limit is around 645V/m).  This means a one meter 
piece of wire that isn't properly grounded can develop 105V of RF energy.  I 
have suffered RF burns of appreciable intensity touching wires that weren't 
connected to anything on either end -- they were just oriented along the 
field gradient.

Also, due to the processing typical radio stations use in the air chain, in 
many cases the quality of playback and record hardware for on-air use must be 
up to the top quality of a recording studio.  Typical radio stations use 
sophisticated compressors, multiband limiters, and many other DSP-driven 
techniques to maximize loudness -- and when the dynamics of the source 
material go pianissimo, the compressors drive up the noise floor to 
compensate.  Noise in the outputs of a broadcast automation or 
music-on-hard-drive system is verboten.  This is particularly (and somewhat 
paradoxically) true on an AM station, where the processing will be much more 
aggressive -- the higher the average modulation, the greater the effective 
coverage area for an AM station.

Having said that, the distortion percentages aren't nearly as important for 
the exact same reason.  But it is wrong to state that a recording studio will 
automatically need higher quality than a radio station.

But in theory I agree with you on the topic of analog I/O into the PC -- but I 
have just seen top quality hardware with on-board analog I/O that had no 
measureable PC-induced noise too many times to ignore.
-- 
Lamar Owen
WGCR Internet Radio
1 Peter 4:11



Re: [linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-27 Thread Paul Davis

immune to the PC's internal noise.  But it all goes to proper design.  The 
Antex SX series of cards likewise contain internal A/D converters.  I have 
measured the noise figures of the Antex SX-36 we have at WGCR using state of 
the art audio systems analyzers, and the noise floor is below the LSB 
threshold.  All due to balanced I/O, sound PC layout techniques, and top of 
the line components.

Just FYI, RME make all the same claims on their website about their 8
channel analog daughterboards for the Hammerfall. 

In a high RF environment, unless the converters are optically isolated from 
the PC, you might be asking for trouble.  When I say 'high RF' I'm talking 
10KW of AM transmitter fifteen feet away, with a measured RF field intensity 
of 105V/m (the ANSI exposure limit is around 645V/m).  This means a one meter 
piece of wire that isn't properly grounded can develop 105V of RF energy.  I 
have suffered RF burns of appreciable intensity touching wires that weren't 
connected to anything on either end -- they were just oriented along the 
field gradient.

Just Say Fibre Optics :) Who needs AES or MADI when you've got ADAT? :)

--p



Re: [linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-25 Thread Mark Rages

On Fri, May 24, 2002 at 10:11:05PM -0400, Paul Davis wrote:
  I don't know what balanced drivers are.
 
 Balanced I/O.  Professional grade.
 
 its related to the wiring of the connectors and the voltage level that
 corresponds to 0dB. footnote: this has nothing to do with drivers
 in the software sense :)

OK, that was my confusion. I know balanced cabling -- I used to work in a 
radio station. The world would be a better place if all audio was 
balanced. Especially car audio. 

Sorry, got off-topic. So: I'm going to resume work on my Gadget Labs
driver soon. Anyone have one of their cards and wants to help?

Regards,
Mark Rages
http://mark.rages.net



Re: [linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-24 Thread Mark Rages

On Fri, May 24, 2002 at 05:47:07PM +1200, Eliot Blennerhassett wrote:
 Hello,
 
 I work for AudioScience (www.audioscience.com) 
 
 We make excellent (how could I say otherwise) audio cards.
 ... 
 I'd like some idea how hard it would be to write an ALSA driver either as a 
 compatibility layer on top of our existing driver, or from the ground up.  I 
 realise that this is rather a broad question, so please consider this an 
 invitation to enter discussion, rather than a request for you to go off and do 
 a lot of work for me.
 

You should be on the alsa-dev mailing list. Writing a driver is probably 
about as easy as giving the card and a copy of the Windows driver source 
to the right person. Otherwise, read the existing drivers for examples.

 Oh - what do you think of the cards' feature set?  
 
 
 Some distinctive things about our cards (not all have all features)
 - they have on board DSP.  Code is downloaded by the driver.

Common.

 - they have a lot of on board buffer memory (hundreds of K at least)

Hopefully that doesn't hurt latency.

 - on board DSP handles decompression/compression

Why? Which formats?

 - mixing

Common on consumer cards.

 - samplerate conversion or multiple outputs at different rates

Interesting. Is there an application for this?

 - analog and digital audio I/O, balanced drivers

I don't know what balanced drivers are.

Regards,
Mark



Re: [linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-24 Thread Fred Gleason

On Friday 24 May 2002 01:47, Eliot Blennerhassett wrote:

 I'd like some idea how hard it would be to write an ALSA driver either as a
 compatibility layer on top of our existing driver, or from the ground up. 
 I realise that this is rather a broad question, so please consider this an
 invitation to enter discussion, rather than a request for you to go off and
 do a lot of work for me.

Howdy Eliot, wonderful to see you on the list!

The two basic components that would have to be addressed to get ALSA support 
are the kernel and user-space (alsa-lib) components.  My sense is that the 
alsa-lib interface could be implemented as a layer between the application 
and HPI, much as the existing Windows sound driver is implemented.  The 
bigger challenge would be to get the kernel part working.  ALSA developers 
are encouraged to use alsa-lib whenever possible, but it *is* possible to use 
the kernel ioctls directly too.  Whether it would make any sense (or would 
even be possible) to implement alsa-lib emulation on top of a different set 
of ioctls is a good question.  

As for supporting features on the cards, other than a few oddball functions 
(like the GPIO calls or HPI_SetStreamOutVelocity()), I don't think there 
would be a great deal of difficulty.  It would be nice to have an envrionment 
where *both* APIs were available -- HPI is quite handy for doing some things.

Cheers!


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Re: [linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-24 Thread Alexander Ehlert

Hi,

 I'd like some idea how hard it would be to write an ALSA driver either 
 as a
 compatibility layer on top of our existing driver, or from the ground 
 up.  I
 realise that this is rather a broad question, so please consider this an
 invitation to enter discussion, rather than a request for you to go off 
 and do
 a lot of work for me.

With that amount of knowledge you have about your card, it actually 
shouldn't
be to hard to develop a driver. I wrote a linux kernel driver for a 
volume rendering
graphics card with reconfigurable hardware and an onboard DSP with less 
information.
I don't know about the interna of your HPI, but surely you need to 
include some low level
code to do the PCI detection and setup of your card with linux. If 
that's already in your
HPI it shouldn't be a problem to adapt that to write a wrapper for the 
alsa lib.
If you want then you provide your dsp code and distribute it with alsa. 
At least some amount of code
goes into the kernel and so has to meet Linus standards :-)

Alright, then. If you have any questions, I see no problem supporting 
you, writing that driver.

Cheers, Alex




Re: [linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-24 Thread Lamar Owen

On Friday 24 May 2002 01:12 pm, Mark Rages wrote:
 On Fri, May 24, 2002 at 05:47:07PM +1200, Eliot Blennerhassett wrote:
  - analog and digital audio I/O, balanced drivers

 I don't know what balanced drivers are.

Balanced I/O.  Professional grade.

As to the AudioScience cards, they are common in radio stations and recording 
studios.  The audio specs (frequency response, distortion) are nearly the 
best in the business.  They are NOT cheap, though.  But multiple balanced ins 
and outs on XLR's (by way of octopus cabling) works very well.

One of the radio stations I engineer for has a couple of the higher end ASI 
cards 4336 maybe?  Anyway, they have the digital I/O (for triggers and 
switching, not digital audio), and they are very happy with the quality of 
the cards.

If these cards get good Linux drivers, this will be killer.
-- 
Lamar Owen
WGCR Internet Radio
1 Peter 4:11



Re: [linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-24 Thread Paul Davis

 I don't know what balanced drivers are.

Balanced I/O.  Professional grade.

its related to the wiring of the connectors and the voltage level that
corresponds to 0dB. footnote: this has nothing to do with drivers
in the software sense :)

As to the AudioScience cards, they are common in radio stations and recording 
studios.  The audio specs (frequency response, distortion) are nearly the 
best in the business.  They are NOT cheap, though.  But multiple balanced ins 
and outs on XLR's (by way of octopus cabling) works very well.

By my standards, anybody who puts A-D/D-A conversion inside their
computer chassis is by definition not concerned with the best in the
business. The term professional audio for me doesn't include analog
I/O for a computer card unless the converters are in a separate
box. It might be good enough for a radio station, but for a recording
studio?

I think you're always better off with 3rd party converters myself, so
you can make you're own choices on the quality level, and possibly
have multiple options (Fostex,Frontier Designs,Apogee for example) as
well as upgrade/downgrade paths.

--p



[linux-audio-dev] Writing a driver for this card: your thoughts?

2002-05-23 Thread Eliot Blennerhassett

Hello,

I work for AudioScience (www.audioscience.com) 

We make excellent (how could I say otherwise) audio cards.

The emphasis within the company has been on microsoft windows drivers.
... but we have a Linux driver, currently proprietary, closed source, that 
exposes this (http://www.audioscience.com/internet/download/spchpi.pdf) API.

(It may be possible to release the host side code, but never the DSP code on 
the cards.)

I think it would be much better if we had an ALSA driver.

I'd like some idea how hard it would be to write an ALSA driver either as a 
compatibility layer on top of our existing driver, or from the ground up.  I 
realise that this is rather a broad question, so please consider this an 
invitation to enter discussion, rather than a request for you to go off and do 
a lot of work for me.

Oh - what do you think of the cards' feature set?  


Some distinctive things about our cards (not all have all features)
- they have on board DSP.  Code is downloaded by the driver.
- they have a lot of on board buffer memory (hundreds of K at least)
- on board DSP handles decompression/compression
- mixing
- samplerate conversion or multiple outputs at different rates
- analog and digital audio I/O, balanced drivers

=
Eliot Blennerhassett   *:-{)
AudioScience, Inc. (New Zealand Office)
6 Centaurus Rd 
Christchurch 8002  Mobile: +64 21 1183531
New ZealandPh/fax: +64  3 3327818
[EMAIL PROTECTED]
http://www.audioscience.com
=