More than one audio codec on a PC mobo

On Thu, 27 Feb 2014 09:22:31 -0800 (PST), Lasse Langwadt Christensen
<langwadt@fonz.dk> wrote:

Den torsdag den 27. februar 2014 02.29.47 UTC+1 skrev k...@attt.bizz:
On Wed, 26 Feb 2014 16:53:56 -0800 (PST), Lasse Langwadt Christensen

langwadt@fonz.dk> wrote:



Den torsdag den 27. februar 2014 01.42.23 UTC+1 skrev Joerg:

bitrex wrote:



On 2/26/2014 2:42 PM, bitrex wrote:



On 2/25/2014 5:01 PM, Joerg wrote:



Folks,







The AC97 standard describes only up to four sound chips operated



simultaneously, on page 21:







ftp://download.intel.com/support/motherboards/desktop/sb/ac97_r23.pdf







What if one would like to connect, say, 20 of them and all are supposed



to run nicely synchronous? Like in a digital mixer board for music.











Here's a dirt cheap way to get a ton of analog audio inputs:







Buy one of these:







http://www.ebay.com/itm/M-Audio-Profire-Lightbridge-/201042188530?pt=US_Computer_Recording_Interfaces&hash=item2ecf0c58f2















4 ADAT lightpipe in/outs







And then get 4 of these:







http://www.soundonsound.com/sos/jun04/articles/behringerada.htm







Then you have 32 analog inputs to Firewire, all synced sample-accurate



via ADAT clock. If you need more, buy another Lightbridge and sync both



setups via their Word Cock connectors.







I neglected to ask if the requirement for 20+ channels of synced audio



to the computer was for a one-off installation, or was a requirement for



some kind of product that you're developing. If it's the latter I'm



curious as to what the application is, because as shown multitrack audio



recording to the PC is a completely solved problem with commodity-priced



hardware already available.











It's not a one-off but for a product. Can't talk about the application



but essentially it's the processing of electrical signals that (luckily)



happen to be spectrally in the audio band. Phase synchronicity of all



channels to each other and dynamic range are the key parameters.





a good start would be to feed them all from the same oscillator, that should take care of the hardest problem, getting the sample rate exactly the same

though there might a pll for some sample rates..



PLLs aren't necessary. This isn't a huge problem at all. Digital

systems tend to be synchronous. ;-)

yes when they run off the same clock, but if you use multiple ADC each with their own PLL to do sample rates that isn't a nice fraction of the xtal rate I'm not so sure

The point being, that the ADCs don't need PLLs at all. The bit/word
clocks keep the samples synchronized. The master clock doesn't really
need to be synchronized. He's only got 40-50 channels. I can do 48
channels in three (six is a little cheaper) DACs and no more than six
ADCs. That's not a huge footprint to route a couple of 12.288MHz
clocks around, if you insist on synchronizing master clocks.

 

On Wed, 26 Feb 2014 17:38:03 -0800, Joerg <invalid@invalid.invalid>
wrote:

Lasse Langwadt Christensen wrote:
Den torsdag den 27. februar 2014 01.42.23 UTC+1 skrev Joerg:
bitrex wrote:

On 2/26/2014 2:42 PM, bitrex wrote:
On 2/25/2014 5:01 PM, Joerg wrote:
Folks,
The AC97 standard describes only up to four sound chips operated
simultaneously, on page 21:
ftp://download.intel.com/support/motherboards/desktop/sb/ac97_r23.pdf
What if one would like to connect, say, 20 of them and all are supposed
to run nicely synchronous? Like in a digital mixer board for music.
Here's a dirt cheap way to get a ton of analog audio inputs:
Buy one of these:
http://www.ebay.com/itm/M-Audio-Profire-Lightbridge-/201042188530?pt=US_Computer_Recording_Interfaces&hash=item2ecf0c58f2
4 ADAT lightpipe in/outs
And then get 4 of these:
http://www.soundonsound.com/sos/jun04/articles/behringerada.htm
Then you have 32 analog inputs to Firewire, all synced sample-accurate
via ADAT clock. If you need more, buy another Lightbridge and sync both
setups via their Word Cock connectors.
I neglected to ask if the requirement for 20+ channels of synced audio
to the computer was for a one-off installation, or was a requirement for
some kind of product that you're developing. If it's the latter I'm
curious as to what the application is, because as shown multitrack audio
recording to the PC is a completely solved problem with commodity-priced
hardware already available.




It's not a one-off but for a product. Can't talk about the application

but essentially it's the processing of electrical signals that (luckily)

happen to be spectrally in the audio band. Phase synchronicity of all

channels to each other and dynamic range are the key parameters.


a good start would be to feed them all from the same oscillator, that should take care of the hardest problem, getting the sample rate exactly the same
though there might a pll for some sample rates..



Feding the same clock is easy but something must make sure that the
samples are all kept in time-sync. Even when they dump their data in
sccessive order.

That's easy. As long as you can generate the data coherently,
delivering it to DACs is a piece of cake. A sample time is forever.

 

[Joerg]

krw@attt.bizz wrote:

On Wed, 26 Feb 2014 17:38:03 -0800, Joerg <invalid@invalid.invalid
wrote:

Lasse Langwadt Christensen wrote:
Den torsdag den 27. februar 2014 01.42.23 UTC+1 skrev Joerg:
bitrex wrote:

[...]

It's not a one-off but for a product. Can't talk about the application

but essentially it's the processing of electrical signals that (luckily)

happen to be spectrally in the audio band. Phase synchronicity of all

channels to each other and dynamic range are the key parameters.

a good start would be to feed them all from the same oscillator, that should take care of the hardest problem, getting the sample rate exactly the same
though there might a pll for some sample rates..


Feding the same clock is easy but something must make sure that the
samples are all kept in time-sync. Even when they dump their data in
sccessive order.

That's easy. As long as you can generate the data coherently,
delivering it to DACs is a piece of cake. A sample time is forever.

I am not concerned about DACs but about ADCs. There is a massive amount
of data. Could be handled by modern USB if there'd be chips and drivers.

--
Regards, Joerg

http://www.analogconsultants.com/

 

[Joerg]

krw@attt.bizz wrote:

On Wed, 26 Feb 2014 17:35:40 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Wed, 26 Feb 2014 16:29:55 -0800, Joerg <invalid@invalid.invalid
wrote:

[...]

I am not sure yet but somewhere around 30-40 audio inputs. Outputs
probably a well but inputs are more important.
With that many, the problem is getting enough TDM channels. Most DACs
only do TDM-8 so that's at least four or five channels. You're also
starting to talk about a significant amount of data. An FPGA might be
in order. What sampling rate do you need? 48kHz? 96kHz?

48kHz would be fine. And yes, this could mean an FPGA. It would just be
nice if it doesn't have to be a total roll-our-own project.

I think if you want to do it for a reasonable price, it's going to go
that way.

Yup, looks like it :-(

[...]

Yes, the DAC part is probably easier because everyone has 5.1 or 7.1
chips. But only two audio-in channels per chip, usually.
Four-channel CODECs are readily available. IIRC, six channel are
around, too. I find separating them to be easier, though if they're
all line level, perhaps not.

There is the PCM3168 with 6-ch input, maybe that can be coaxed into a
synchronous orchestra with others.

The level can be handled via amplifiers, that's the easy part.

My point was that depending on the power level needed, separate DACs
and ADCs can be easier than CODECs (keeping the ADCs away from power
amps).

Sure, but then things become quite expensive if you want 18 bits or
more. Sound codecs have the mass market advantage when it comes to pricing.

--
Regards, Joerg

http://www.analogconsultants.com/

 

On Thu, 27 Feb 2014 17:37:53 -0800, Joerg <invalid@invalid.invalid>
wrote:

krw@attt.bizz wrote:
On Wed, 26 Feb 2014 17:38:03 -0800, Joerg <invalid@invalid.invalid
wrote:

Lasse Langwadt Christensen wrote:
Den torsdag den 27. februar 2014 01.42.23 UTC+1 skrev Joerg:
bitrex wrote:


[...]

It's not a one-off but for a product. Can't talk about the application

but essentially it's the processing of electrical signals that (luckily)

happen to be spectrally in the audio band. Phase synchronicity of all

channels to each other and dynamic range are the key parameters.

a good start would be to feed them all from the same oscillator, that should take care of the hardest problem, getting the sample rate exactly the same
though there might a pll for some sample rates..


Feding the same clock is easy but something must make sure that the
samples are all kept in time-sync. Even when they dump their data in
sccessive order.

That's easy. As long as you can generate the data coherently,
delivering it to DACs is a piece of cake. A sample time is forever.


I am not concerned about DACs but about ADCs. There is a massive amount
of data. Could be handled by modern USB if there'd be chips and drivers.

It's the same problem, but backwards. ;-) Synchronization isn't the
problem. You don't have *that* many channels.

 

On Thu, 27 Feb 2014 17:18:48 -0800, Joerg <invalid@invalid.invalid>
wrote:

ftp://download.intel.com/support/motherboards/desktop/sb/ac97_r23.pdf

What if one would like to connect, say, 20 of them and all are supposed
to run nicely synchronous? Like in a digital mixer board for music.

What exactly are you trying to achieve ?

Create a 20 channel audio mixer ?

Sampling some 20 discrete channels ? At what bandwidth (sampling rate)
and at what accuracy (number of bits or SNR) do you need ?


Can't reveal details but it's 20+ electrical signals that spectrally
happen to be in the audio range. Several kHz. Sampling range needs to be
40kHz and the achievable SNR should be well north of 90dB.

What are your latency requirements ?

While that number of channels and SNR requirements, should be doable
with a PCI card inside a computer, but for more demanding
requirements, I would look at some external ADC board(s), which would
simplify (balanced) signal conditioning, EMC issues and galvanic
isolation issues.

With an external board with say 8 channels and an ethernet interface
might be an alternative.

With three bytes/sample that would generate 24 bytes/sample. With two
samples/frame this would nicely fill the minimum size ethernet frame.
With a few more samples/frame and the bit efficiency would be even
better, allowing several such eight channel units even through a
single 100BaseT ethernet port on the host. Larger systems or higher
sampling rates would require a 1000BaseT connection between the
ethernet switch and the host computer, while the ADC board to switch
connections could still be 100BaseT.

To synchronize the sample clocks on different ADC units, some NTP
style system could be used.

 

On Thu, 27 Feb 2014 17:35:29 -0800, Joerg <invalid@invalid.invalid>
wrote:

krw@attt.bizz wrote:
On Wed, 26 Feb 2014 17:35:40 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Wed, 26 Feb 2014 16:29:55 -0800, Joerg <invalid@invalid.invalid
wrote:


[...]


I am not sure yet but somewhere around 30-40 audio inputs. Outputs
probably a well but inputs are more important.
With that many, the problem is getting enough TDM channels. Most DACs
only do TDM-8 so that's at least four or five channels. You're also
starting to talk about a significant amount of data. An FPGA might be
in order. What sampling rate do you need? 48kHz? 96kHz?

48kHz would be fine. And yes, this could mean an FPGA. It would just be
nice if it doesn't have to be a total roll-our-own project.

I think if you want to do it for a reasonable price, it's going to go
that way.


Yup, looks like it :-(

[...]


Yes, the DAC part is probably easier because everyone has 5.1 or 7.1
chips. But only two audio-in channels per chip, usually.
Four-channel CODECs are readily available. IIRC, six channel are
around, too. I find separating them to be easier, though if they're
all line level, perhaps not.

There is the PCM3168 with 6-ch input, maybe that can be coaxed into a
synchronous orchestra with others.

The level can be handled via amplifiers, that's the easy part.

My point was that depending on the power level needed, separate DACs
and ADCs can be easier than CODECs (keeping the ADCs away from power
amps).


Sure, but then things become quite expensive if you want 18 bits or
more. Sound codecs have the mass market advantage when it comes to pricing.

Nah. >110dB dynamic range differential output DACs are cheap. We pay
something under a buck a piece for 8-channel DACs. Our quantities are
higher than you're likely to see, though. ;-) At 1K volumes, they're
probably three or four bucks.

 

[Joerg]

krw@attt.bizz wrote:

On Thu, 27 Feb 2014 17:35:29 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Wed, 26 Feb 2014 17:35:40 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Wed, 26 Feb 2014 16:29:55 -0800, Joerg <invalid@invalid.invalid
wrote:

[...]


I am not sure yet but somewhere around 30-40 audio inputs. Outputs
probably a well but inputs are more important.
With that many, the problem is getting enough TDM channels. Most DACs
only do TDM-8 so that's at least four or five channels. You're also
starting to talk about a significant amount of data. An FPGA might be
in order. What sampling rate do you need? 48kHz? 96kHz?

48kHz would be fine. And yes, this could mean an FPGA. It would just be
nice if it doesn't have to be a total roll-our-own project.
I think if you want to do it for a reasonable price, it's going to go
that way.

Yup, looks like it :-(

[...]


Yes, the DAC part is probably easier because everyone has 5.1 or 7.1
chips. But only two audio-in channels per chip, usually.
Four-channel CODECs are readily available. IIRC, six channel are
around, too. I find separating them to be easier, though if they're
all line level, perhaps not.
There is the PCM3168 with 6-ch input, maybe that can be coaxed into a
synchronous orchestra with others.

The level can be handled via amplifiers, that's the easy part.
My point was that depending on the power level needed, separate DACs
and ADCs can be easier than CODECs (keeping the ADCs away from power
amps).

Sure, but then things become quite expensive if you want 18 bits or
more. Sound codecs have the mass market advantage when it comes to pricing.

Nah. >110dB dynamic range differential output DACs are cheap. We pay
something under a buck a piece for 8-channel DACs. Our quantities are
higher than you're likely to see, though. ;-) At 1K volumes, they're
probably three or four bucks.

Again, it's about ADCs here. DACs are a piece of cake.

--
Regards, Joerg

http://www.analogconsultants.com/

 

[Joerg]

krw@attt.bizz wrote:

On Thu, 27 Feb 2014 17:37:53 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Wed, 26 Feb 2014 17:38:03 -0800, Joerg <invalid@invalid.invalid
wrote:

Lasse Langwadt Christensen wrote:
Den torsdag den 27. februar 2014 01.42.23 UTC+1 skrev Joerg:
bitrex wrote:

[...]

It's not a one-off but for a product. Can't talk about the application

but essentially it's the processing of electrical signals that (luckily)

happen to be spectrally in the audio band. Phase synchronicity of all

channels to each other and dynamic range are the key parameters.

a good start would be to feed them all from the same oscillator, that should take care of the hardest problem, getting the sample rate exactly the same
though there might a pll for some sample rates..

Feding the same clock is easy but something must make sure that the
samples are all kept in time-sync. Even when they dump their data in
sccessive order.
That's easy. As long as you can generate the data coherently,
delivering it to DACs is a piece of cake. A sample time is forever.

I am not concerned about DACs but about ADCs. There is a massive amount
of data. Could be handled by modern USB if there'd be chips and drivers.

It's the same problem, but backwards. ;-) Synchronization isn't the
problem. You don't have *that* many channels.

Well, I will have that many channels. I am just looking for a possible
cheapo solution using sound chips. If there is no viable one then I'll
do it the classic way. BTDT, the most massive one had 64 ADCs.

--
Regards, Joerg

http://www.analogconsultants.com/

 

[Joerg]

upsidedown@downunder.com wrote:

On Thu, 27 Feb 2014 17:18:48 -0800, Joerg <invalid@invalid.invalid
wrote:

ftp://download.intel.com/support/motherboards/desktop/sb/ac97_r23.pdf

What if one would like to connect, say, 20 of them and all are supposed
to run nicely synchronous? Like in a digital mixer board for music.
What exactly are you trying to achieve ?

Create a 20 channel audio mixer ?

Sampling some 20 discrete channels ? At what bandwidth (sampling rate)
and at what accuracy (number of bits or SNR) do you need ?

Can't reveal details but it's 20+ electrical signals that spectrally
happen to be in the audio range. Several kHz. Sampling range needs to be
40kHz and the achievable SNR should be well north of 90dB.

What are your latency requirements ?

Latency does not matter, provided it is exactly the same for all ADC
channels or (if not) the data is time-stamped.

While that number of channels and SNR requirements, should be doable
with a PCI card inside a computer, but for more demanding
requirements, I would look at some external ADC board(s), which would
simplify (balanced) signal conditioning, EMC issues and galvanic
isolation issues.

They are usually prohibitively expensive.

With an external board with say 8 channels and an ethernet interface
might be an alternative.

Yes, but at lab equipment pricing levels

With three bytes/sample that would generate 24 bytes/sample. With two
samples/frame this would nicely fill the minimum size ethernet frame.
With a few more samples/frame and the bit efficiency would be even
better, allowing several such eight channel units even through a
single 100BaseT ethernet port on the host. Larger systems or higher
sampling rates would require a 1000BaseT connection between the
ethernet switch and the host computer, while the ADC board to switch
connections could still be 100BaseT.

To synchronize the sample clocks on different ADC units, some NTP
style system could be used.

I'd rather go in via PCI or USB, it is probably simpler and we could
likely pull a lot from FPGA vendor libraries.

--
Regards, Joerg

http://www.analogconsultants.com/

 

On Fri, 28 Feb 2014 07:36:17 -0800, Joerg <invalid@invalid.invalid>
wrote:

krw@attt.bizz wrote:
On Thu, 27 Feb 2014 17:35:29 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Wed, 26 Feb 2014 17:35:40 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Wed, 26 Feb 2014 16:29:55 -0800, Joerg <invalid@invalid.invalid
wrote:

[...]


I am not sure yet but somewhere around 30-40 audio inputs. Outputs
probably a well but inputs are more important.
With that many, the problem is getting enough TDM channels. Most DACs
only do TDM-8 so that's at least four or five channels. You're also
starting to talk about a significant amount of data. An FPGA might be
in order. What sampling rate do you need? 48kHz? 96kHz?

48kHz would be fine. And yes, this could mean an FPGA. It would just be
nice if it doesn't have to be a total roll-our-own project.
I think if you want to do it for a reasonable price, it's going to go
that way.

Yup, looks like it :-(

[...]


Yes, the DAC part is probably easier because everyone has 5.1 or 7.1
chips. But only two audio-in channels per chip, usually.
Four-channel CODECs are readily available. IIRC, six channel are
around, too. I find separating them to be easier, though if they're
all line level, perhaps not.
There is the PCM3168 with 6-ch input, maybe that can be coaxed into a
synchronous orchestra with others.

The level can be handled via amplifiers, that's the easy part.
My point was that depending on the power level needed, separate DACs
and ADCs can be easier than CODECs (keeping the ADCs away from power
amps).

Sure, but then things become quite expensive if you want 18 bits or
more. Sound codecs have the mass market advantage when it comes to pricing.

Nah. >110dB dynamic range differential output DACs are cheap. We pay
something under a buck a piece for 8-channel DACs. Our quantities are
higher than you're likely to see, though. ;-) At 1K volumes, they're
probably three or four bucks.


Again, it's about ADCs here. DACs are a piece of cake.

ADCs are just backwards DACs. They're a bit more expensive but still
cheap, though I don't know of any 8-channel ones. I use four-channel
ADCs (106dB) all the time, though. Audio stuff is really cheap.

 

On Fri, 28 Feb 2014 07:38:13 -0800, Joerg <invalid@invalid.invalid>
wrote:

krw@attt.bizz wrote:
On Thu, 27 Feb 2014 17:37:53 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Wed, 26 Feb 2014 17:38:03 -0800, Joerg <invalid@invalid.invalid
wrote:

Lasse Langwadt Christensen wrote:
Den torsdag den 27. februar 2014 01.42.23 UTC+1 skrev Joerg:
bitrex wrote:

[...]

It's not a one-off but for a product. Can't talk about the application

but essentially it's the processing of electrical signals that (luckily)

happen to be spectrally in the audio band. Phase synchronicity of all

channels to each other and dynamic range are the key parameters.

a good start would be to feed them all from the same oscillator, that should take care of the hardest problem, getting the sample rate exactly the same
though there might a pll for some sample rates..

Feding the same clock is easy but something must make sure that the
samples are all kept in time-sync. Even when they dump their data in
sccessive order.
That's easy. As long as you can generate the data coherently,
delivering it to DACs is a piece of cake. A sample time is forever.

I am not concerned about DACs but about ADCs. There is a massive amount
of data. Could be handled by modern USB if there'd be chips and drivers.

It's the same problem, but backwards. ;-) Synchronization isn't the
problem. You don't have *that* many channels.


Well, I will have that many channels. I am just looking for a possible
cheapo solution using sound chips. If there is no viable one then I'll
do it the classic way. BTDT, the most massive one had 64 ADCs.

You said forty or fifty. That's not many, when they come four and
eight to the whack. Even 64 isn't that large. Synchronization is
*NOT* your problem. That's the easy part.

 

[Joerg]

krw@attt.bizz wrote:

On Fri, 28 Feb 2014 07:36:17 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Thu, 27 Feb 2014 17:35:29 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Wed, 26 Feb 2014 17:35:40 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Wed, 26 Feb 2014 16:29:55 -0800, Joerg <invalid@invalid.invalid
wrote:

[...]


I am not sure yet but somewhere around 30-40 audio inputs. Outputs
probably a well but inputs are more important.
With that many, the problem is getting enough TDM channels. Most DACs
only do TDM-8 so that's at least four or five channels. You're also
starting to talk about a significant amount of data. An FPGA might be
in order. What sampling rate do you need? 48kHz? 96kHz?

48kHz would be fine. And yes, this could mean an FPGA. It would just be
nice if it doesn't have to be a total roll-our-own project.
I think if you want to do it for a reasonable price, it's going to go
that way.

Yup, looks like it :-(

[...]


Yes, the DAC part is probably easier because everyone has 5.1 or 7.1
chips. But only two audio-in channels per chip, usually.
Four-channel CODECs are readily available. IIRC, six channel are
around, too. I find separating them to be easier, though if they're
all line level, perhaps not.
There is the PCM3168 with 6-ch input, maybe that can be coaxed into a
synchronous orchestra with others.

The level can be handled via amplifiers, that's the easy part.
My point was that depending on the power level needed, separate DACs
and ADCs can be easier than CODECs (keeping the ADCs away from power
amps).

Sure, but then things become quite expensive if you want 18 bits or
more. Sound codecs have the mass market advantage when it comes to pricing.
Nah. >110dB dynamic range differential output DACs are cheap. We pay
something under a buck a piece for 8-channel DACs. Our quantities are
higher than you're likely to see, though. ;-) At 1K volumes, they're
probably three or four bucks.

Again, it's about ADCs here. DACs are a piece of cake.

ADCs are just backwards DACs. They're a bit more expensive but still
cheap, though I don't know of any 8-channel ones. I use four-channel
ADCs (106dB) all the time, though. Audio stuff is really cheap.

Audio yes, regular ADCs no. I just needed one for a design and 12-bit
1-ch at 100ksps is already above a buck. If you want 18-bits or better
anything other than audio is really expensive. DACs are cheap, ADCs aren't.

--
Regards, Joerg

http://www.analogconsultants.com/

 

[Joerg]

krw@attt.bizz wrote:

On Fri, 28 Feb 2014 07:38:13 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Thu, 27 Feb 2014 17:37:53 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Wed, 26 Feb 2014 17:38:03 -0800, Joerg <invalid@invalid.invalid
wrote:

Lasse Langwadt Christensen wrote:
Den torsdag den 27. februar 2014 01.42.23 UTC+1 skrev Joerg:
bitrex wrote:

[...]

It's not a one-off but for a product. Can't talk about the application

but essentially it's the processing of electrical signals that (luckily)

happen to be spectrally in the audio band. Phase synchronicity of all

channels to each other and dynamic range are the key parameters.

a good start would be to feed them all from the same oscillator, that should take care of the hardest problem, getting the sample rate exactly the same
though there might a pll for some sample rates..

Feding the same clock is easy but something must make sure that the
samples are all kept in time-sync. Even when they dump their data in
sccessive order.
That's easy. As long as you can generate the data coherently,
delivering it to DACs is a piece of cake. A sample time is forever.

I am not concerned about DACs but about ADCs. There is a massive amount
of data. Could be handled by modern USB if there'd be chips and drivers.
It's the same problem, but backwards. ;-) Synchronization isn't the
problem. You don't have *that* many channels.

Well, I will have that many channels. I am just looking for a possible
cheapo solution using sound chips. If there is no viable one then I'll
do it the classic way. BTDT, the most massive one had 64 ADCs.

You said forty or fifty. That's not many, when they come four and
eight to the whack. Even 64 isn't that large. Synchronization is
*NOT* your problem. That's the easy part.

In this case it's 40-50, the 64-ch version was a project way back when
and that one had a BW north of 25MHz. Sync for that was not cheap.

I was hoping for something easy like a special big fat USB hub or
something. But I guess it either has to be ADAT or roll-our-own.

--
Regards, Joerg

http://www.analogconsultants.com/

 

On Fri, 28 Feb 2014 15:44:40 -0800, Joerg <invalid@invalid.invalid>
wrote:

krw@attt.bizz wrote:
On Fri, 28 Feb 2014 07:36:17 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Thu, 27 Feb 2014 17:35:29 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Wed, 26 Feb 2014 17:35:40 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Wed, 26 Feb 2014 16:29:55 -0800, Joerg <invalid@invalid.invalid
wrote:

[...]


I am not sure yet but somewhere around 30-40 audio inputs. Outputs
probably a well but inputs are more important.
With that many, the problem is getting enough TDM channels. Most DACs
only do TDM-8 so that's at least four or five channels. You're also
starting to talk about a significant amount of data. An FPGA might be
in order. What sampling rate do you need? 48kHz? 96kHz?

48kHz would be fine. And yes, this could mean an FPGA. It would just be
nice if it doesn't have to be a total roll-our-own project.
I think if you want to do it for a reasonable price, it's going to go
that way.

Yup, looks like it :-(

[...]


Yes, the DAC part is probably easier because everyone has 5.1 or 7.1
chips. But only two audio-in channels per chip, usually.
Four-channel CODECs are readily available. IIRC, six channel are
around, too. I find separating them to be easier, though if they're
all line level, perhaps not.
There is the PCM3168 with 6-ch input, maybe that can be coaxed into a
synchronous orchestra with others.

The level can be handled via amplifiers, that's the easy part.
My point was that depending on the power level needed, separate DACs
and ADCs can be easier than CODECs (keeping the ADCs away from power
amps).

Sure, but then things become quite expensive if you want 18 bits or
more. Sound codecs have the mass market advantage when it comes to pricing.
Nah. >110dB dynamic range differential output DACs are cheap. We pay
something under a buck a piece for 8-channel DACs. Our quantities are
higher than you're likely to see, though. ;-) At 1K volumes, they're
probably three or four bucks.

Again, it's about ADCs here. DACs are a piece of cake.

ADCs are just backwards DACs. They're a bit more expensive but still
cheap, though I don't know of any 8-channel ones. I use four-channel
ADCs (106dB) all the time, though. Audio stuff is really cheap.


Audio yes, regular ADCs no. I just needed one for a design and 12-bit
1-ch at 100ksps is already above a buck. If you want 18-bits or better
anything other than audio is really expensive. DACs are cheap, ADCs aren't.

But you just said you needed to digitize audio. Are you moving the
goal posts?

 

[Joerg]

krw@attt.bizz wrote:

On Fri, 28 Feb 2014 15:44:40 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Fri, 28 Feb 2014 07:36:17 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Thu, 27 Feb 2014 17:35:29 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Wed, 26 Feb 2014 17:35:40 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Wed, 26 Feb 2014 16:29:55 -0800, Joerg <invalid@invalid.invalid
wrote:

[...]


I am not sure yet but somewhere around 30-40 audio inputs. Outputs
probably a well but inputs are more important.
With that many, the problem is getting enough TDM channels. Most DACs
only do TDM-8 so that's at least four or five channels. You're also
starting to talk about a significant amount of data. An FPGA might be
in order. What sampling rate do you need? 48kHz? 96kHz?

48kHz would be fine. And yes, this could mean an FPGA. It would just be
nice if it doesn't have to be a total roll-our-own project.
I think if you want to do it for a reasonable price, it's going to go
that way.

Yup, looks like it :-(

[...]


Yes, the DAC part is probably easier because everyone has 5.1 or 7.1
chips. But only two audio-in channels per chip, usually.
Four-channel CODECs are readily available. IIRC, six channel are
around, too. I find separating them to be easier, though if they're
all line level, perhaps not.
There is the PCM3168 with 6-ch input, maybe that can be coaxed into a
synchronous orchestra with others.

The level can be handled via amplifiers, that's the easy part.
My point was that depending on the power level needed, separate DACs
and ADCs can be easier than CODECs (keeping the ADCs away from power
amps).

Sure, but then things become quite expensive if you want 18 bits or
more. Sound codecs have the mass market advantage when it comes to pricing.
Nah. >110dB dynamic range differential output DACs are cheap. We pay
something under a buck a piece for 8-channel DACs. Our quantities are
higher than you're likely to see, though. ;-) At 1K volumes, they're
probably three or four bucks.
Again, it's about ADCs here. DACs are a piece of cake.
ADCs are just backwards DACs. They're a bit more expensive but still
cheap, though I don't know of any 8-channel ones. I use four-channel
ADCs (106dB) all the time, though. Audio stuff is really cheap.

Audio yes, regular ADCs no. I just needed one for a design and 12-bit
1-ch at 100ksps is already above a buck. If you want 18-bits or better
anything other than audio is really expensive. DACs are cheap, ADCs aren't.

But you just said you needed to digitize audio. Are you moving the
goal posts?

No. I wanted to know how to line up umpteen audio codecs in a way that
they talk to a PC and remain 100% synchronous. Using ADCs for otyher
markets is expensive.

--
Regards, Joerg

http://www.analogconsultants.com/

 

On Fri, 28 Feb 2014 16:18:41 -0800, Joerg <invalid@invalid.invalid>
wrote:

krw@attt.bizz wrote:
On Fri, 28 Feb 2014 15:44:40 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Fri, 28 Feb 2014 07:36:17 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Thu, 27 Feb 2014 17:35:29 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Wed, 26 Feb 2014 17:35:40 -0800, Joerg <invalid@invalid.invalid
wrote:

krw@attt.bizz wrote:
On Wed, 26 Feb 2014 16:29:55 -0800, Joerg <invalid@invalid.invalid
wrote:

[...]


I am not sure yet but somewhere around 30-40 audio inputs. Outputs
probably a well but inputs are more important.
With that many, the problem is getting enough TDM channels. Most DACs
only do TDM-8 so that's at least four or five channels. You're also
starting to talk about a significant amount of data. An FPGA might be
in order. What sampling rate do you need? 48kHz? 96kHz?

48kHz would be fine. And yes, this could mean an FPGA. It would just be
nice if it doesn't have to be a total roll-our-own project.
I think if you want to do it for a reasonable price, it's going to go
that way.

Yup, looks like it :-(

[...]


Yes, the DAC part is probably easier because everyone has 5.1 or 7.1
chips. But only two audio-in channels per chip, usually.
Four-channel CODECs are readily available. IIRC, six channel are
around, too. I find separating them to be easier, though if they're
all line level, perhaps not.
There is the PCM3168 with 6-ch input, maybe that can be coaxed into a
synchronous orchestra with others.

The level can be handled via amplifiers, that's the easy part.
My point was that depending on the power level needed, separate DACs
and ADCs can be easier than CODECs (keeping the ADCs away from power
amps).

Sure, but then things become quite expensive if you want 18 bits or
more. Sound codecs have the mass market advantage when it comes to pricing.
Nah. >110dB dynamic range differential output DACs are cheap. We pay
something under a buck a piece for 8-channel DACs. Our quantities are
higher than you're likely to see, though. ;-) At 1K volumes, they're
probably three or four bucks.
Again, it's about ADCs here. DACs are a piece of cake.
ADCs are just backwards DACs. They're a bit more expensive but still
cheap, though I don't know of any 8-channel ones. I use four-channel
ADCs (106dB) all the time, though. Audio stuff is really cheap.

Audio yes, regular ADCs no. I just needed one for a design and 12-bit
1-ch at 100ksps is already above a buck. If you want 18-bits or better
anything other than audio is really expensive. DACs are cheap, ADCs aren't.

But you just said you needed to digitize audio. Are you moving the
goal posts?


No. I wanted to know how to line up umpteen audio codecs in a way that
they talk to a PC and remain 100% synchronous. Using ADCs for otyher
markets is expensive.

I guess the subject and other posts in the thread just confused me.
;-)

 

On Fri, 28 Feb 2014 16:18:41 -0800, Joerg <invalid@invalid.invalid>
wrote:

No. I wanted to know how to line up umpteen audio codecs in a way that
they talk to a PC and remain 100% synchronous. Using ADCs for otyher
markets is expensive.

You did not say anything about expected production volumes (just a one
off project or thousands of units) and hence is it feasible to design
your own PCI card.

If you have to design your own PCI card, what is wrong with the old
stereo audio ADCs such as the CS5394, apart from the board space
needed for 10+ stereo ADC chips ?

It has differential inputs, a 2 Hz audio high pass (so no DC
measurement) and I˛S interface. It can operated both in master or
slave mode, put one in master mode and the rest in slave mode and tie
all the clock signals together.

Of course you will need a FPGA to convert these I˛S signals serial
signals to something usable for a computer, such as 32 bit parallel
words/sample to presented to the PCI bridge chip.

If some serial interface such as Ethernet, USB or PCIe is used, then
put the sample from every channel into the same frame (preferably 32
bits/sample to simplify CPU load), thus all samples from a particular
time would be guarantied to be available in the same frame.

Then you will have to think how you will get the data from the PC
interface card into memory, perhaps DMA or interrupt driven. However,
if every sample will generate an interrupt, there would be 40000
interrupts each second at 40 kHz sampling rate, which is quite a lot.
Transferring 40 samples for each interrupt would give 1000
interrupts/second, which is a reasonable value.

If the data is to be processed with some soft-RT system, such as
Windows or Linux, you may still need quite a lot of software FIFO
buffering between the ISR and user mode code.

So it is not just the ADC interface you need to consider but also
other parts of the system, in order to avoid making some local
optimizing that greatly harms the rest of the system.

 

[Lasse Langwadt Christense]

Den torsdag den 27. februar 2014 19.01.33 UTC+1 skrev Joerg:

Lasse Langwadt Christensen wrote:

Den torsdag den 27. februar 2014 02.38.03 UTC+1 skrev Joerg:

Lasse Langwadt Christensen wrote:



Den torsdag den 27. februar 2014 01.42.23 UTC+1 skrev Joerg:

bitrex wrote:

On 2/26/2014 2:42 PM, bitrex wrote:

On 2/25/2014 5:01 PM, Joerg wrote:

Folks,

The AC97 standard describes only up to four sound chips operated

simultaneously, on page 21:

ftp://download.intel.com/support/motherboards/desktop/sb/ac97_r23.pdf

What if one would like to connect, say, 20 of them and all are supposed

to run nicely synchronous? Like in a digital mixer board for music.

Here's a dirt cheap way to get a ton of analog audio inputs:

Buy one of these:

http://www.ebay.com/itm/M-Audio-Profire-Lightbridge-/201042188530?pt=US_Computer_Recording_Interfaces&hash=item2ecf0c58f2

4 ADAT lightpipe in/outs

And then get 4 of these:

http://www.soundonsound.com/sos/jun04/articles/behringerada.htm

Then you have 32 analog inputs to Firewire, all synced sample-accurate

via ADAT clock. If you need more, buy another Lightbridge and sync both

setups via their Word Cock connectors.

I neglected to ask if the requirement for 20+ channels of synced audio

to the computer was for a one-off installation, or was a requirement for

some kind of product that you're developing. If it's the latter I'm

curious as to what the application is, because as shown multitrack audio

recording to the PC is a completely solved problem with commodity-priced

hardware already available.

It's not a one-off but for a product. Can't talk about the application

but essentially it's the processing of electrical signals that (luckily)

happen to be spectrally in the audio band. Phase synchronicity of all

channels to each other and dynamic range are the key parameters.

a good start would be to feed them all from the same oscillator, that should take care of the hardest problem, getting the sample rate exactly the same

though there might a pll for some sample rates..









Feding the same clock is easy but something must make sure that the



samples are all kept in time-sync. Even when they dump their data in



sccessive order.





I understand



could you occasionally feed all the inputs from a calibration source?

sorta like a clapper board





Yes, that is possible but painful because it requires extra muxes.

might only require one if there is a way to just add it to the inputs
say if you have a virtual ground one all the input amps, mux that between a reference signal and your virtual ground potential

if you want to get fancy something like a prbs to get a nice correlation

peak



must it be done real time?





Not really but close. If there is a delay of a few seconds that's ok but

there will be signals coming in all the time, continuously.

ok, so you'll be processing it in real time, I'd think that an fpga or MCU
with USB giving you a nice stream of all you channels would be less hassle than trying to get the whole windows/linux audio system to play nice with multiple channels

something like a small FPGA for sync and de-serializing and an FT2232H
for USB. One channel for configuring the FPGA, other as parallel FIFO

-Lasse

 

On Sat, 01 Mar 2014 09:15:55 +0200, upsidedown@downunder.com wrote:

On Fri, 28 Feb 2014 16:18:41 -0800, Joerg <invalid@invalid.invalid
wrote:

No. I wanted to know how to line up umpteen audio codecs in a way that
they talk to a PC and remain 100% synchronous. Using ADCs for otyher
markets is expensive.

You did not say anything about expected production volumes (just a one
off project or thousands of units) and hence is it feasible to design
your own PCI card.

If you have to design your own PCI card, what is wrong with the old
stereo audio ADCs such as the CS5394, apart from the board space
needed for 10+ stereo ADC chips ?

It has differential inputs, a 2 Hz audio high pass (so no DC
measurement) and I˛S interface. It can operated both in master or
slave mode, put one in master mode and the rest in slave mode and tie
all the clock signals together.

Yep. I'd use something larger than stereo and some version or TDM,
rather than I2S but that's just details. I'd need to know more about
the application before diving too far into the details.

Of course you will need a FPGA to convert these I˛S signals serial
signals to something usable for a computer, such as 32 bit parallel
words/sample to presented to the PCI bridge chip.

Precisely. You could pull the PCI bridge into the FPGA but I'd
probably go the easy way, too. The PLX bridges were a piece of cake
to use. That was a *long* time ago, though. ;-)

If some serial interface such as Ethernet, USB or PCIe is used, then
put the sample from every channel into the same frame (preferably 32
bits/sample to simplify CPU load), thus all samples from a particular
time would be guarantied to be available in the same frame.

Sure, but it's not a big deal to buffer them, either. You just need
to make sure that the buffers can be kept full. Like I said,
synchronizing the hardware is the easy part.

Then you will have to think how you will get the data from the PC
interface card into memory, perhaps DMA or interrupt driven. However,
if every sample will generate an interrupt, there would be 40000
interrupts each second at 40 kHz sampling rate, which is quite a lot.
Transferring 40 samples for each interrupt would give 1000
interrupts/second, which is a reasonable value.

Sure. Again, I'd buffer the samples at the bridge to keep the
interrupts under control.

If the data is to be processed with some soft-RT system, such as
Windows or Linux, you may still need quite a lot of software FIFO
buffering between the ISR and user mode code.

Do it in hardware, at the I2S/TDM controller. Dual-port memory on
FPGAs is pretty cheap (once you've bought into an FPGA).

So it is not just the ADC interface you need to consider but also
other parts of the system, in order to avoid making some local
optimizing that greatly harms the rest of the system.

Yep. The hardware is the easy part. ;-)