There's a path to follow!
Peufeu
has described his extreme approach to an audiophile pc system with enough diy content 
to keep anyone challenged for more than a few weekends. Go see his site for some very good reading! It's a big influence on my intentions.
Many are curious whether you have measured the resulting dynamic range and jitter you are getting out of your dac or through your digital interface to the dac?
Regarding losing bits to volume control - yes, but...
Foobar2000, for example, is currently converting my 16-bit sound file information into 24-bit fixed point and padding to 32 bits. Assuming the M-Audio dac is receiving 24 bits from foobar, there are 8 empty bits to use for shifting the words right to attenuate volume. I'll try to find out something on this unless someone already knows the answer. Anyway, just like the Wadia does for volume control, it doesn't matter much if you lose empty bits.
-Robert
Peufeu
has described his extreme approach to an audiophile pc system with enough diy content to keep anyone challenged for more than a few weekends. Go see his site for some very good reading! It's a big influence on my intentions.Many are curious whether you have measured the resulting dynamic range and jitter you are getting out of your dac or through your digital interface to the dac?
Regarding losing bits to volume control - yes, but...
Foobar2000, for example, is currently converting my 16-bit sound file information into 24-bit fixed point and padding to 32 bits. Assuming the M-Audio dac is receiving 24 bits from foobar, there are 8 empty bits to use for shifting the words right to attenuate volume. I'll try to find out something on this unless someone already knows the answer. Anyway, just like the Wadia does for volume control, it doesn't matter much if you lose empty bits.
-Robert
Hmmm????
Having trouble finding any info at all out there on how foobar does the volume control.
And where are the jitter comparisons from the vast pool of experts on this forum? I would have expected some really impressive sub-100ps system specs coming in from the transport-oriented people.
Please, if this data is sitting in some thread I've missed, just point us to it!!
Did you think the Stereophile characterization of the RME card was not relevant or was it impressive? Is this type of characterization totally irrelevant to most of you?
-Robert
Having trouble finding any info at all out there on how foobar does the volume control.
And where are the jitter comparisons from the vast pool of experts on this forum? I would have expected some really impressive sub-100ps system specs coming in from the transport-oriented people.
Please, if this data is sitting in some thread I've missed, just point us to it!!
Did you think the Stereophile characterization of the RME card was not relevant or was it impressive? Is this type of characterization totally irrelevant to most of you?
-Robert
Digital volume control
You say losing empty bits doesn't matter...
If the DAC chip were perfect, this would be the case. However I'd rather use my DAC at its full resolution and attenuate it afterwards.
With seal multibit DACs (PCM1704) I don't know if it is a problem, but with sigma-delta DACs it probably is, as these usually don't have 24 real bits of resolution ; their resolution comes from averaging...
Anyway I haven't compared the sonics so I'm not going to try to force one particular solution down your throat.
About the RME Digi96, I've been surprised to hear how good it sounds. Certainly better than the same price in CD players, and you get 8 channels. It's not as good as a top notch DAC, of course, but hey, that's pretty normal considering the price.
I wish I had more time to play with these things right now. So much work, I have to watch you get all the fun !
You say losing empty bits doesn't matter...
If the DAC chip were perfect, this would be the case. However I'd rather use my DAC at its full resolution and attenuate it afterwards.
With seal multibit DACs (PCM1704) I don't know if it is a problem, but with sigma-delta DACs it probably is, as these usually don't have 24 real bits of resolution ; their resolution comes from averaging...
Anyway I haven't compared the sonics so I'm not going to try to force one particular solution down your throat.
About the RME Digi96, I've been surprised to hear how good it sounds. Certainly better than the same price in CD players, and you get 8 channels. It's not as good as a top notch DAC, of course, but hey, that's pretty normal considering the price.
I wish I had more time to play with these things right now. So much work, I have to watch you get all the fun !
adat
I also think the PC solution offers many many advantages which make it very worthwhile even if not on par with world class transports/cd plavers/DACs, but even that is a matter of controversy if done right. The noise of the PC is not an issue in my opinion, for you can move it to the next room or place it in the closet if you have a sufficiently big one in your listening room.
I`m planning to use my PC with an moderately priced ADAT soundcard connected to 3 non-os DACs (6-channels) using TDA1543s and 6 Gainclones to power 3 way stereo speakers, digital crossovers & dsp via Brutefir or perhaps (better) some Windows based solution if thers is one available for both tasks (hints?). Two Seagate 160gb HDs (& raid) would secure me from traumatic data loss and give enoung storage if a lossless compression format is used.
Peu-feu:
Do you know any sources in Europe for the ADAT Alesis chip? Can you point me to a site where an application such as the one you mention (clock feedback, ADAT) is described in detail?
Can you suggest any adat soundcards?
Regards,
Sebastian
I also think the PC solution offers many many advantages which make it very worthwhile even if not on par with world class transports/cd plavers/DACs, but even that is a matter of controversy if done right. The noise of the PC is not an issue in my opinion, for you can move it to the next room or place it in the closet if you have a sufficiently big one in your listening room.
I`m planning to use my PC with an moderately priced ADAT soundcard connected to 3 non-os DACs (6-channels) using TDA1543s and 6 Gainclones to power 3 way stereo speakers, digital crossovers & dsp via Brutefir or perhaps (better) some Windows based solution if thers is one available for both tasks (hints?). Two Seagate 160gb HDs (& raid) would secure me from traumatic data loss and give enoung storage if a lossless compression format is used.
Peu-feu:
Do you know any sources in Europe for the ADAT Alesis chip? Can you point me to a site where an application such as the one you mention (clock feedback, ADAT) is described in detail?
Can you suggest any adat soundcards?
Regards,
Sebastian
I might do the same
Sebastian,
Your project sounds potentially interesting to me. When I'm finished with my 2-way analog-out tests (probably late this summer), I might do something a lot like you're planning.
I'm definitely going to set up a master low-jitter clock scheme for the soundcard and multiple dacs.
One thing that's starting to intrigue me is to buy a soundcard and strip it clean of power supply(ies), clock, interface hardware and so on and then build it up with super-regulated supplies and great op-amps, etc, then mount it in a "clean case" with good control of emi. Perhaps a usb version would be more amenable; don't know yet.
The whole thing, pc and amps and power supplies will all be inside a custom-made case. It would be great if someone could convince Peter Daniel to start building pc music systems soon, so we can see what they should look like and how they should be constructed. Peter???
On the almost-pc-music news-scene, did everybody see the April/May issue of TAS on page 52 where an iPod connected by RadioShack patch cable thru a Parasound amp driving Wilson's Sophia speakers surpised everyone at CES and beat-out a Krell/Krell/big-$-unnamed-speaker combo in a head-to-head. This was staged by Dave Wilson, whose point was that speakers are really important, but my take is that the iPod may have some potential for mod'ing that I hadn't considered. I wonder.
-Robert
Sebastian,
Your project sounds potentially interesting to me. When I'm finished with my 2-way analog-out tests (probably late this summer), I might do something a lot like you're planning.
I'm definitely going to set up a master low-jitter clock scheme for the soundcard and multiple dacs.
One thing that's starting to intrigue me is to buy a soundcard and strip it clean of power supply(ies), clock, interface hardware and so on and then build it up with super-regulated supplies and great op-amps, etc, then mount it in a "clean case" with good control of emi. Perhaps a usb version would be more amenable; don't know yet.
The whole thing, pc and amps and power supplies will all be inside a custom-made case. It would be great if someone could convince Peter Daniel to start building pc music systems soon, so we can see what they should look like and how they should be constructed. Peter???
On the almost-pc-music news-scene, did everybody see the April/May issue of TAS on page 52 where an iPod connected by RadioShack patch cable thru a Parasound amp driving Wilson's Sophia speakers surpised everyone at CES and beat-out a Krell/Krell/big-$-unnamed-speaker combo in a head-to-head. This was staged by Dave Wilson, whose point was that speakers are really important, but my take is that the iPod may have some potential for mod'ing that I hadn't considered. I wonder.
-Robert
Re: adat
This is exactly what im doing/done.
Im half way there atm using brutefir for the active xovers, but using the analog outs from the sound card to an 8ch bridged gainclone amp.
Ill start building a 6ch DAC when i have time.
You should think about buying one more 160gb (three), so you can setup RAID5. That way you only loose 1/3 of total capacity instead of 1/2 when using RAID1.
swak said:
This is exactly what im doing/done.
Im half way there atm using brutefir for the active xovers, but using the analog outs from the sound card to an 8ch bridged gainclone amp.
Ill start building a 6ch DAC when i have time.
You should think about buying one more 160gb (three), so you can setup RAID5. That way you only loose 1/3 of total capacity instead of 1/2 when using RAID1.
Re: I might do the same
If you sunc your soudcard to the master clock in the DAC, you won't need to dissect it... just feed it a clock signal...
However, this restricts you to one master clock (and of course, one DAC) but you can stuff as many DAC chips in it as you want.
Also your DAC can output a word clock (just a square wave at the right frequency, really) which you feed to the soundcard and slave it to it.
I started by using a CD723 and slaving it to the master clock in the DAC simply by using a cable and pulse transformer. However I wouldn't advise you to do that because sending 8 MHz in a cable can generate a lot of radiation if you don't take care of matching impedances perfectly. I know mine radiates and I will fix this in the next version.
The CD723 can be used as a CD player but I don't use it because it is crippled (it loses the LSB) ; however it produces a SPDIF stream in sync with the master clock, which I use to slave my soundcard on.
I could have used a SPDIF encodre chip instead of a CD723 if I had thought about this in the beginning... this will be for the next version !
There will be a slight lag between the master clock and the signal sent by the soundcard (roundtrip and processing delay), but the CS8412 has a double buffer receiver which takes care of that ; and it outputs signal which is full of jitter but in sync with the master clock inside the DAC. It never complains about missed samples and it looks perfct on the scope.
When I will redo it, I'll fix the following :
- use completely optical teansmission between computer and DAC, both ways (clock and data)
- use a format which allows 24-192 x 8 channels. ADAT does NOT allow that (16-44 x 8 max) so I'd have to use a Hammerfall fard with several ADAT channels in interleave mode (very easy to do with Alsa on Linux) and some PLD in my DAC to put the bits together. Or I'll use an USB chip if I can find one with these high data rates (will have to be USB 2.0 I guess !)
My RME DAC soundcard has a special power supply connector which you plug in an unused Floppy drive supply ; thus the power comes straight from the PSU and doesn't go through the noisy motherboard. RME makes digital-only soundcards (which feed from the PCI bus) and a DAC board, which you put in the PCI slot just to hold it in place, but it hasn't got any copper on the connector. It is simply powered via the floppy cable, and connected to the digital soundcard with an ADAT cable. You could put this card outside the computer.
However, thses cards have multilayer PCBs (mine has ground planes on both sides, you don't see any tracks, they are all inside, I suspect 4 layers), and it's all microscopic SMD's, so good luck for modding them !
Have fun !
Pierre
If you sunc your soudcard to the master clock in the DAC, you won't need to dissect it... just feed it a clock signal...
However, this restricts you to one master clock (and of course, one DAC) but you can stuff as many DAC chips in it as you want.
Also your DAC can output a word clock (just a square wave at the right frequency, really) which you feed to the soundcard and slave it to it.
I started by using a CD723 and slaving it to the master clock in the DAC simply by using a cable and pulse transformer. However I wouldn't advise you to do that because sending 8 MHz in a cable can generate a lot of radiation if you don't take care of matching impedances perfectly. I know mine radiates and I will fix this in the next version.
The CD723 can be used as a CD player but I don't use it because it is crippled (it loses the LSB) ; however it produces a SPDIF stream in sync with the master clock, which I use to slave my soundcard on.
I could have used a SPDIF encodre chip instead of a CD723 if I had thought about this in the beginning... this will be for the next version !
There will be a slight lag between the master clock and the signal sent by the soundcard (roundtrip and processing delay), but the CS8412 has a double buffer receiver which takes care of that ; and it outputs signal which is full of jitter but in sync with the master clock inside the DAC. It never complains about missed samples and it looks perfct on the scope.
When I will redo it, I'll fix the following :
- use completely optical teansmission between computer and DAC, both ways (clock and data)
- use a format which allows 24-192 x 8 channels. ADAT does NOT allow that (16-44 x 8 max) so I'd have to use a Hammerfall fard with several ADAT channels in interleave mode (very easy to do with Alsa on Linux) and some PLD in my DAC to put the bits together. Or I'll use an USB chip if I can find one with these high data rates (will have to be USB 2.0 I guess !)
My RME DAC soundcard has a special power supply connector which you plug in an unused Floppy drive supply ; thus the power comes straight from the PSU and doesn't go through the noisy motherboard. RME makes digital-only soundcards (which feed from the PCI bus) and a DAC board, which you put in the PCI slot just to hold it in place, but it hasn't got any copper on the connector. It is simply powered via the floppy cable, and connected to the digital soundcard with an ADAT cable. You could put this card outside the computer.
However, thses cards have multilayer PCBs (mine has ground planes on both sides, you don't see any tracks, they are all inside, I suspect 4 layers), and it's all microscopic SMD's, so good luck for modding them !
Have fun !
Pierre
Why use S/PDIF?
I don't want to start any great debates, but being a computer we have many options for output. Not just S/PDIF. I agree that any form of analogue output from inside the PC - which is inherently noisey due to the very high frequency components inside will have trouble matching the sound quality of a decent CD transport. Although i'm sure you could make it very good, by using some of the ideas on this thread.
But CD transports are stuck with S/PDIF outputs, a protocol that depends on synchronous communications to reassemble the audio - errors will always introduced, even after extensive reclocking. These defeciencies have been known for a while, but the popularity and ease of use of S/PDIF have continued its use.
http://www.enjoythemusic.com/magazine/viewpoint/0401/deficienciesofspdif.htm
has some good arguments on S/PDIF.
So why use a synchronous datastream from a PC when we can use an asynchronous stream, buffer and clock the data out at the DAC end using a precision clock source. Therefore totally eliminanting the inherent problems with S/PDIF and infact creating a 'better' transport than any CD could - jitter (clock sync errors on a synchronous link) will always have an effect further down the line. An asynchronous link - as long as the data is fed quick enough - doesn't have an effect at the source and so only the quality of the clock source and DAC at the analogue end will have an effect.
I would really like to hear peoples opinions on this.
Thanks
Andrew
P.S. a silent PC can be bought for far cheaper than many CD transports.....
I don't want to start any great debates, but being a computer we have many options for output. Not just S/PDIF. I agree that any form of analogue output from inside the PC - which is inherently noisey due to the very high frequency components inside will have trouble matching the sound quality of a decent CD transport. Although i'm sure you could make it very good, by using some of the ideas on this thread.
But CD transports are stuck with S/PDIF outputs, a protocol that depends on synchronous communications to reassemble the audio - errors will always introduced, even after extensive reclocking. These defeciencies have been known for a while, but the popularity and ease of use of S/PDIF have continued its use.
http://www.enjoythemusic.com/magazine/viewpoint/0401/deficienciesofspdif.htm
has some good arguments on S/PDIF.
So why use a synchronous datastream from a PC when we can use an asynchronous stream, buffer and clock the data out at the DAC end using a precision clock source. Therefore totally eliminanting the inherent problems with S/PDIF and infact creating a 'better' transport than any CD could - jitter (clock sync errors on a synchronous link) will always have an effect further down the line. An asynchronous link - as long as the data is fed quick enough - doesn't have an effect at the source and so only the quality of the clock source and DAC at the analogue end will have an effect.
I would really like to hear peoples opinions on this.
Thanks
Andrew
P.S. a silent PC can be bought for far cheaper than many CD transports.....
Asynchronous
You cound use a full asynchronous transfer with an USB chip (they are available) but you won't get many channels because USB 1.1 is bandwidth-limited... I dunno if this would do 24-192
If there is a USB2.0 or Firewire chip, now things could get going !
What you suggest is putting all the logic (including buffering) inside the DAC.
What I did is simply split this and use an already made buffering & asynchronous logic, which is in my PCI soundcard, and have it communicate with my DAC synchronously (the master clock being fed by the DAC).
I agree this is not the most beautiful solution, but it was the simplest to implement, as SPDIF decoders are readily available, and SPDIF is very reliable for data transfer, the only thing it doesn't get right is the clock !
I'd prefer having a USB2 chip and stick it in my DAC though... this would be more elegant and I could use all this bandwidth for high sample rates and bitwidths, to do the oversampling in the PC with custom filters... I'd like to try that...
I'm open to all suggestions !
You cound use a full asynchronous transfer with an USB chip (they are available) but you won't get many channels because USB 1.1 is bandwidth-limited... I dunno if this would do 24-192
If there is a USB2.0 or Firewire chip, now things could get going !
What you suggest is putting all the logic (including buffering) inside the DAC.
What I did is simply split this and use an already made buffering & asynchronous logic, which is in my PCI soundcard, and have it communicate with my DAC synchronously (the master clock being fed by the DAC).
I agree this is not the most beautiful solution, but it was the simplest to implement, as SPDIF decoders are readily available, and SPDIF is very reliable for data transfer, the only thing it doesn't get right is the clock !
I'd prefer having a USB2 chip and stick it in my DAC though... this would be more elegant and I could use all this bandwidth for high sample rates and bitwidths, to do the oversampling in the PC with custom filters... I'd like to try that...
I'm open to all suggestions !
S/PDIF Clock
Hi, i hear what you are saying about the badnwidth and i agree that it is ashame that there are no readily available USB2 audio codecs, but the oversampling can be done inside the DAC - like it would be on the other end of a CD transport, but the point is that the source quality would no longer make any difference. No need to send much time / money on getting the source signal right, its digital and transferred asynchronously so it makes no difference by definition. I'm trying to say that there is no physical reason why a PC can not be a better digital audio source than a CD transport, simply because it suffers no jitter.
"SPDIF actually has to be synched to the exact frequency of the transport (i.e. if the transport is working at say 44.0896K instead of 44.1K the dac has to sync to that frequency). Therefore the jitter problems of SPDIF almost go away using USB."
Infact modifying a present DAC design by replacing the S/PDIF receiver with a USB to I2S receiver and removing the clock regeneration circuitry should improve the sound quality.
Does anyone know of a USB to I2S that allows the output to be clocked from a precision clock source? This could then drive the DAC chip directly and there would be no need for any clock regeneration and other jitter control circuitry.
Any ideas?
Andrew
Hi, i hear what you are saying about the badnwidth and i agree that it is ashame that there are no readily available USB2 audio codecs, but the oversampling can be done inside the DAC - like it would be on the other end of a CD transport, but the point is that the source quality would no longer make any difference. No need to send much time / money on getting the source signal right, its digital and transferred asynchronously so it makes no difference by definition. I'm trying to say that there is no physical reason why a PC can not be a better digital audio source than a CD transport, simply because it suffers no jitter.
"SPDIF actually has to be synched to the exact frequency of the transport (i.e. if the transport is working at say 44.0896K instead of 44.1K the dac has to sync to that frequency). Therefore the jitter problems of SPDIF almost go away using USB."
Infact modifying a present DAC design by replacing the S/PDIF receiver with a USB to I2S receiver and removing the clock regeneration circuitry should improve the sound quality.
Does anyone know of a USB to I2S that allows the output to be clocked from a precision clock source? This could then drive the DAC chip directly and there would be no need for any clock regeneration and other jitter control circuitry.
Any ideas?
Andrew
On the look-out
USB 2.0 external totally asynch diy soundcard approach seems like it's worth looking into further. I googled around and still haven't found such an interface ic although it's obvious things like this are possible since there are so many usb 2.0 soundcards now.
I wonder if we could get an i2s feed from any of the audio chips on modern motherboards? This would bypass usb entirely and would also be 100% compatible with software drivers already in place.
BTW I'm working hard on a new gainclone amp for my pc atm and so haven't been spending much time on this thread in the last few days. But I'm excited about getting to it soon!
-Robert
USB 2.0 external totally asynch diy soundcard approach seems like it's worth looking into further. I googled around and still haven't found such an interface ic although it's obvious things like this are possible since there are so many usb 2.0 soundcards now.
I wonder if we could get an i2s feed from any of the audio chips on modern motherboards? This would bypass usb entirely and would also be 100% compatible with software drivers already in place.
BTW I'm working hard on a new gainclone amp for my pc atm and so haven't been spending much time on this thread in the last few days. But I'm excited about getting to it soon!
-Robert
MWP said:
First of all, very interesting and educating thread. Secondly in response to the above:
I currently use a Via Epia mini-itx board (see mini-itx.com ). The board is about 6x6 and is fanless. Also the power supply is mounted externally in the form of a brick psu.
These mini-itx boards are a great way to get started in your audio pc venture. The cpu is onboard and many models are fanless.
As far as players go foobar is great and I use Steinberg's wavelab to rip audio CDs.
I2S
I2S is NOT desirable.
I2S leaves the master clock in the PC. Granted, it's better than SPDIF because there is no mixing of clock and data, but the clock is still generated in a noisy environment and has to travel along a cable, possible opto-isolators and such, to the DAC...
And SPDIF is cheaper, because a SPDIF transformer + CS8141 = about 20 euros, while hi-speed opto-isolators for I2S are very expensive !
I'd definitely rather have USB 2 !
By the way, how do you insulate an USB2 connection ? Transformers ??????
I2S is NOT desirable.
I2S leaves the master clock in the PC. Granted, it's better than SPDIF because there is no mixing of clock and data, but the clock is still generated in a noisy environment and has to travel along a cable, possible opto-isolators and such, to the DAC...
And SPDIF is cheaper, because a SPDIF transformer + CS8141 = about 20 euros, while hi-speed opto-isolators for I2S are very expensive !
I'd definitely rather have USB 2 !
By the way, how do you insulate an USB2 connection ? Transformers ??????
Usb2
Check this out.
http://www.semiconductors.philips.com/pip/ISP1581BD.html
"The ISP1581 is a cost-optimized and feature-optimized Hi-Speed Universal Serial Bus (USB) interface device, which fully complies with the Universal Serial Bus Specification Rev. 2.0. It provides high-speed USB communication capacity to systems based on a microcontroller or microprocessor. The ISP1581 communicates with the system’s microcontroller/processor through a high-speed general-purpose parallel interface."
Check this out.
http://www.semiconductors.philips.com/pip/ISP1581BD.html
"The ISP1581 is a cost-optimized and feature-optimized Hi-Speed Universal Serial Bus (USB) interface device, which fully complies with the Universal Serial Bus Specification Rev. 2.0. It provides high-speed USB communication capacity to systems based on a microcontroller or microprocessor. The ISP1581 communicates with the system’s microcontroller/processor through a high-speed general-purpose parallel interface."
General USB2.0 interfaces
peufeu,
There are many interfaces similar to the philips one you linked. It provides the hardware function we want but leaves a problem with lack of drivers.
Hmmm. Are you suggesting we simply output the music file (.wav, .flac or whatever) through usb to an external device (our dac) and then let our dac take control of all decoding? Be careful or our dac will turn into a pc
Or do you want to write new drivers for our application?
-Robert
peufeu,
There are many interfaces similar to the philips one you linked. It provides the hardware function we want but leaves a problem with lack of drivers.
Hmmm. Are you suggesting we simply output the music file (.wav, .flac or whatever) through usb to an external device (our dac) and then let our dac take control of all decoding? Be careful or our dac will turn into a pc
Or do you want to write new drivers for our application?
-Robert
Re: General USB2.0 interfaces
Hum... let's hit googoole :
http://www.semiconductors.philips.com/buses/usb/software_downloads/
Good one :
http://www.beyondlogic.org/usb/usbhard2.htm
ATMEL makes one !
http://www.eeglossary.com/usb.htm
open source drivers everywhere... anyways coding something wouldn't scare me... providing it's on Linux and not on Windows...
We could even put a small microcontroller in the DAC which would behave as an IDE device, and connect it to a 29 euro USB2 compact flash reader... no need for drivers... I'm pretty sure this kind of ATA code already exists in the open source community... and it seems the ATA protocol is pretty simple, too.
Then we'd just write data as raw blocks with some linux command like dd ! It's very simple to make Linux believe a soundcard is a file and vice versa. You don't even have to do anything special.
It's just an idea...
Well no. I want the PC do do ALL the processing, including Oversampling/dithering/room correction (with BruteFIR) and the DAC to be as dumb as possible. Just take the bits and play them, sort of. I also want complete electrical insulation. And the master clock in the DAC, of course !
However it looks like the simplest solution will still involve some intelligence and microcontroller hacking, not to mention impossible to get, impossible to solder, USB chips...
RFScheer said:
Hum... let's hit googoole :
http://www.semiconductors.philips.com/buses/usb/software_downloads/
Good one :
http://www.beyondlogic.org/usb/usbhard2.htm
ATMEL makes one !
http://www.eeglossary.com/usb.htm
open source drivers everywhere... anyways coding something wouldn't scare me... providing it's on Linux and not on Windows...
We could even put a small microcontroller in the DAC which would behave as an IDE device, and connect it to a 29 euro USB2 compact flash reader... no need for drivers... I'm pretty sure this kind of ATA code already exists in the open source community... and it seems the ATA protocol is pretty simple, too.
Then we'd just write data as raw blocks with some linux command like dd ! It's very simple to make Linux believe a soundcard is a file and vice versa. You don't even have to do anything special.
It's just an idea...
Well no. I want the PC do do ALL the processing, including Oversampling/dithering/room correction (with BruteFIR) and the DAC to be as dumb as possible. Just take the bits and play them, sort of. I also want complete electrical insulation. And the master clock in the DAC, of course !
However it looks like the simplest solution will still involve some intelligence and microcontroller hacking, not to mention impossible to get, impossible to solder, USB chips...
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