Hello,
I have recently been pondering DIY adc's- I notice there are much more resources for Dac's- I have a recording studio and It would be awesome to put together some high end ADC's for much less then buying them- Plus the satisfaction of DIY is always nice!!
At $25 each this is far less then the units that use these same components- even if you factor in all the other stuff you need! A benchmark, or mytek or other unit that uses these same chips for an 8 channel ADC runs about 4k- I could probably do 24 channels for that much-
I was looking at this one from cirrus logic (crystal semiconductor)-
http://www.cirrus.com/en/products/pro/detail/P1024.html
it is the model cs5381-
I guess what I was wondering was if anyone can point me in the direction of a site that has the basics of how to go about doing this??
Also what about sample clock- would you guys trust the internal clock on these babies or use somthing else?
And how complicated is it to get a AES/EBU or spdif output from one of these packages?
And- are all ADC's created for line level audio? I've been thinking about the possibility of having an ADC at mic level- with a stepped attenuator- or a pot to prevent clipping- Kinda like the $10,000 Nuemann digital mic-
Most mics are very low level 150ohm output from the balancing transformer-
Anyway thanks for any input / sites that I can learn more from,
Ryan
I have recently been pondering DIY adc's- I notice there are much more resources for Dac's- I have a recording studio and It would be awesome to put together some high end ADC's for much less then buying them- Plus the satisfaction of DIY is always nice!!
At $25 each this is far less then the units that use these same components- even if you factor in all the other stuff you need! A benchmark, or mytek or other unit that uses these same chips for an 8 channel ADC runs about 4k- I could probably do 24 channels for that much-
I was looking at this one from cirrus logic (crystal semiconductor)-
http://www.cirrus.com/en/products/pro/detail/P1024.html
it is the model cs5381-
I guess what I was wondering was if anyone can point me in the direction of a site that has the basics of how to go about doing this??
Also what about sample clock- would you guys trust the internal clock on these babies or use somthing else?
And how complicated is it to get a AES/EBU or spdif output from one of these packages?
And- are all ADC's created for line level audio? I've been thinking about the possibility of having an ADC at mic level- with a stepped attenuator- or a pot to prevent clipping- Kinda like the $10,000 Nuemann digital mic-
Most mics are very low level 150ohm output from the balancing transformer-
Anyway thanks for any input / sites that I can learn more from,
Ryan
take a look at this pdf file, it is a devboard from akm.
you will see how they get a line level input and a mike level input as well.
http://www.asahi-kasei.co.jp/akm/en/product/ak5394a/ekd5394a.pdf
another thing: be careful about the mike you want to use while some needs a "phantom power supply" to run.
bye
you will see how they get a line level input and a mike level input as well.
http://www.asahi-kasei.co.jp/akm/en/product/ak5394a/ekd5394a.pdf
another thing: be careful about the mike you want to use while some needs a "phantom power supply" to run.
bye
RyanC said:
This topic crops up from time to time and then dies out without any real progress as you'll find if you do a search.
Only if you plan to do the pcb yourself and opt to avoid the more exotic aspects like level and status display , multichannel optical output, higher sample rates and mono AES, multiple sync source selection etc. and just have 12 spdif outputs.
The evaluation board schematic provides a turnkey solution and
covers the basics of how to connect the analogue stage, A/D converter and SPDIF transmitter.
What it will not cover is clock generation and distribution,locking to external sources and a few other clock related issues.
They all have a maximum input value and so long as your preceding circuitry stops you exceeding this, you are pretty much free to do what you want.
Any questions, call Guido Tent.
Cool-
It would be hard to get a setup that would support 12 spdif inputs to a computer so that would be an issue if I were to want to go with 24 channels. I actually did search for "ADC" on this forum and found nothing- but I will dig a little deaper-
Is there a DIY solution for the tascam TDIF multichannel format? I wonder if it is really any different then 8 channels of spdif on a db25 cable?? The output from the convertor itself is Spdif right? but will only support 96k as 192 requires one spdif cable per channel. (this is fine by me as I have no intent on recording at 192)
Anyway lets say I need no level display- I can see that on the computer. BUT I do want to do higher sample rates (at least 96k)- Also in the white paper stuff it looks like the CS5381 will self clock if no clock is recieved- Obviosly this is problematic in a bigger system (like mine) but just somthing I noticed.
So is it's input just a standard Wordclock? What if I got a commercially available master clock like the aardsync II. This could be used to control a bunch of cs5381's.
I was looking at the schematics for the input buffers and it seems like that is the hard part- It uses a differential input, is that any different than balanced line level- it doesnt seem like it- what would happen with no input buffer at all- just connected directly to a balanced line level output of a micpreamp-
It would be hard to get a setup that would support 12 spdif inputs to a computer so that would be an issue if I were to want to go with 24 channels. I actually did search for "ADC" on this forum and found nothing- but I will dig a little deaper-
Is there a DIY solution for the tascam TDIF multichannel format? I wonder if it is really any different then 8 channels of spdif on a db25 cable?? The output from the convertor itself is Spdif right? but will only support 96k as 192 requires one spdif cable per channel. (this is fine by me as I have no intent on recording at 192)
Anyway lets say I need no level display- I can see that on the computer. BUT I do want to do higher sample rates (at least 96k)- Also in the white paper stuff it looks like the CS5381 will self clock if no clock is recieved- Obviosly this is problematic in a bigger system (like mine) but just somthing I noticed.
So is it's input just a standard Wordclock? What if I got a commercially available master clock like the aardsync II. This could be used to control a bunch of cs5381's.
I was looking at the schematics for the input buffers and it seems like that is the hard part- It uses a differential input, is that any different than balanced line level- it doesnt seem like it- what would happen with no input buffer at all- just connected directly to a balanced line level output of a micpreamp-
Ryan, this is not really that hard if you don't need a complex analog front end (limiters, etc) but I doubt you currently have the kind of knowledge to implement a project like this from scratch. The CS5381 will NOT output S/PDIF, you need a S/PDIF converter IC like CS8405a et al.
If you have time and money to spend, I suggest you design your stereo analog front end, ADC, and a basic S/PDIF transmitter on a small PCB with lots of test points. Proto it at bare bones PCB (get at least a few copies) and make sure it works. Do an ADAT or other proprietary multichannel board too if you are confident.
Now fix the mistakes. If you've designed the stereo test boards intelligently, you can chain them to proto your multichannel concept. You may even get away with using a slew of slave boards with a master clock/transmitter controlling them. This would be much less prone to error than designing a 24-channel board and hoping it all works.
Before you do this, do a lot of reading. Make sure the chips you use speak compatible formats and logic levels. And lots of test points! (You will use these later to rewire your prototypes.)
If you have time and money to spend, I suggest you design your stereo analog front end, ADC, and a basic S/PDIF transmitter on a small PCB with lots of test points. Proto it at bare bones PCB (get at least a few copies) and make sure it works. Do an ADAT or other proprietary multichannel board too if you are confident.
Now fix the mistakes. If you've designed the stereo test boards intelligently, you can chain them to proto your multichannel concept. You may even get away with using a slew of slave boards with a master clock/transmitter controlling them. This would be much less prone to error than designing a 24-channel board and hoping it all works.
Before you do this, do a lot of reading. Make sure the chips you use speak compatible formats and logic levels. And lots of test points! (You will use these later to rewire your prototypes.
)RyanC said:
Never seen the spec for TDIF and I don't know if there is a chipset easily available and ADAT won't support 96k as standard, so if you want to avoid interfaces that would take years to develop then AES/EBU on a 25way D will probably be your best bet. It also has the advantage of supporting 96k and 192k in single and dual AES formats.
You might want to consider bringing the overflow LED's to the front panel
Assuming the A/D chip is in slave mode, it would require 3 clock signals MCLK,SCLK and LRCLK. Typically the frequencies are 256Fs, 64Fs and Fs respectively, where Fs is the sample rate though the multipliers can and do vary between different A/D chips and sample rates.
A standard SPG like the Aardsync can be used but it can only provide a wordclock reference signal. The MCLK for the A/D chip still has to be generated onboard and it is this onboard clock that is synchronized to wordclock usually with a VCXO.
It only looks hard and is probably the simplest part. If you do not need variable gain at the input there are one chip solutions available.
You might want to consider buying an evaluation board and then work up to synchronizing it to an external source.
Yea thanks guys-
It seems like the way to go is with an evaluation board- The top of the line AKM chip looks a little beter then the CS one-
I'm wondering what the purpose of the input buffer is?? If I don't need any gain- That can be easily controlled from the mic pre- And I already have a balanced signal- again from the mic pre- then what does it do?? I guess what I am wondering is- Isn't a mic preamp essentially an input buffer?
I saw some people on this forum discussing using only a sowter tranny as an output buffer- That would be similer in some respects to my mic pre's- (except it does have gain)- The mic pre puts out balanced +4
I'm sure that there is somthing I'm missing here- and I am sure that you guys know why I need the input buffer- I'm just tying to understand a little more-
True on the overload LED's thats not a bad idea- at the same time it is still not necessary- I use Nuendo and you can see the waveform as your recording- really I find this to be a better meter then anything else-
True on the ADAT thing- Although I would hazard to guess that any modern adat transmitter supports the S/MUX protocall for 4ch 96k operation- My MOTU 2408 MKiii supports it even though they make no mention of it any any documentation- I would still rather not mess with that-
The problem with the AES thing is that soundcards that support it are rare- actually I only know of one the the newest lynx card- Wich does not have enough foldback mixing features to support my needs (for different headphone mixes).
What would work on the AKM side is the optical spdif out put chip- The Motu that I have now supports switching the ADAT's for optical lightpipe- there are 3 sets on a 2408 and the system supports 4 2408's so that could give me 24ch. at some point- besides I don't need to start with 24 channels- 2 seems alot more practical!
Anyway the evaluation boards seem like a good way to go- where is a good source for the AKM ones? The AK5394A looks to be really nice-
Thanks again guys- I really appriciate your patience with an idiot like me!!!-
Ryan
It seems like the way to go is with an evaluation board- The top of the line AKM chip looks a little beter then the CS one-
I'm wondering what the purpose of the input buffer is?? If I don't need any gain- That can be easily controlled from the mic pre- And I already have a balanced signal- again from the mic pre- then what does it do?? I guess what I am wondering is- Isn't a mic preamp essentially an input buffer?
I saw some people on this forum discussing using only a sowter tranny as an output buffer- That would be similer in some respects to my mic pre's- (except it does have gain)- The mic pre puts out balanced +4
I'm sure that there is somthing I'm missing here- and I am sure that you guys know why I need the input buffer- I'm just tying to understand a little more-
True on the overload LED's thats not a bad idea- at the same time it is still not necessary- I use Nuendo and you can see the waveform as your recording- really I find this to be a better meter then anything else-
True on the ADAT thing- Although I would hazard to guess that any modern adat transmitter supports the S/MUX protocall for 4ch 96k operation- My MOTU 2408 MKiii supports it even though they make no mention of it any any documentation- I would still rather not mess with that-
The problem with the AES thing is that soundcards that support it are rare- actually I only know of one the the newest lynx card- Wich does not have enough foldback mixing features to support my needs (for different headphone mixes).
What would work on the AKM side is the optical spdif out put chip- The Motu that I have now supports switching the ADAT's for optical lightpipe- there are 3 sets on a 2408 and the system supports 4 2408's so that could give me 24ch. at some point- besides I don't need to start with 24 channels- 2 seems alot more practical!
Anyway the evaluation boards seem like a good way to go- where is a good source for the AKM ones? The AK5394A looks to be really nice-
Thanks again guys- I really appriciate your patience with an idiot like me!!!-
Ryan
Read the datasheet. The primary goals of the input stage would be to condition the signal so that the ADC can recieve it--a MIC pre could well be all the input circutry you need provided it has the appropriate sensitivity to drive your ADC full range (and no farther!)
The problem with 4ch/96k ADAT would be you need either an IC or some circuitry to drive the 8-channel transmitter with 4 mono channels. I'm guessing all you need is a couple of flip flops that alternately sample and hold each bit for 2-cycles to produce 2 48kHz streams from 1 96kHz one. Needless to say, this sounds like trouble.
The problem with 4ch/96k ADAT would be you need either an IC or some circuitry to drive the 8-channel transmitter with 4 mono channels. I'm guessing all you need is a couple of flip flops that alternately sample and hold each bit for 2-cycles to produce 2 48kHz streams from 1 96kHz one. Needless to say, this sounds like trouble.
TDIF Converter
Hi
I've read the above about a DIY TDIF converter, and it looks pretty good. I'm wondering if anybody has a schematic for an 8 channel TDIF encoder and decoder, that runs at at least 24bit 48kHz.
I understand that TDIF is a simmilar format to Roland RBUS, just having different pin-outs, so therefore a circuit for TDIF could be used with my Presonus V-Fire TDIF Interface...
Is it also the same format as SPDIF? If this is the case then a simple diagram for a SPDIF converter (to above specs) would work too.
Thanks in advance!
Guy Joseph
Hi
I've read the above about a DIY TDIF converter, and it looks pretty good. I'm wondering if anybody has a schematic for an 8 channel TDIF encoder and decoder, that runs at at least 24bit 48kHz.
I understand that TDIF is a simmilar format to Roland RBUS, just having different pin-outs, so therefore a circuit for TDIF could be used with my Presonus V-Fire TDIF Interface...
Is it also the same format as SPDIF? If this is the case then a simple diagram for a SPDIF converter (to above specs) would work too.
Thanks in advance!
Guy Joseph
I've made an ADC around the TI PCM4202 chip and it works really well.
A little bit different to what you're doing, however.
I've got an M Audio Revolution7.1 sound card that is very user friendly for the DIYer.
The card itself routes I2S to all the on board converter chips, so it's not that hard of a task to utilise all of these. In other words the VIA PCI interface chip handles all of the clock generation.
At the bottom of my PC there's a PCB I've built containing a PCM1794 and the 4202, for DAC and ADC conversion. I simply hook them up to the correct I2S lines from the sound card and away I go.
It's powered by super regulators, with the transformer + rectifiers + smoothing caps in a box external to the computer case and sounds excellent.
I have no idea how the recording industry works, but I'm surprised at all of the ADCs and SPDIF stuff being thrown around, as you've described. I'd have thought everything would remain analogue, right up to the final conversion, which would be done through one high quality converter inside some elaborate sound card.
Most of the TI datasheets are fairly good at explaining how to use their chips, and if any special requirements are needed, it tends to list them too.
A little bit different to what you're doing, however.
I've got an M Audio Revolution7.1 sound card that is very user friendly for the DIYer.
The card itself routes I2S to all the on board converter chips, so it's not that hard of a task to utilise all of these. In other words the VIA PCI interface chip handles all of the clock generation.
At the bottom of my PC there's a PCB I've built containing a PCM1794 and the 4202, for DAC and ADC conversion. I simply hook them up to the correct I2S lines from the sound card and away I go.
It's powered by super regulators, with the transformer + rectifiers + smoothing caps in a box external to the computer case and sounds excellent.
I have no idea how the recording industry works, but I'm surprised at all of the ADCs and SPDIF stuff being thrown around, as you've described. I'd have thought everything would remain analogue, right up to the final conversion, which would be done through one high quality converter inside some elaborate sound card.
Most of the TI datasheets are fairly good at explaining how to use their chips, and if any special requirements are needed, it tends to list them too.
All,
A few weeks ago I did order the A/D converter kit from Uwe Beis
'AD24QS - Audio Analog to Digital Converter 24 Bit / 192 kHz'
Check it out at:
http://www.beis.de/Elektronik/ADDA24QS/AD24QS.html
My kit will be available in this month, and I am going to use it for my balanced(input) riaa preamp. As soon as I have it running I will lett you all know
Regards,
Frans.
A few weeks ago I did order the A/D converter kit from Uwe Beis
'AD24QS - Audio Analog to Digital Converter 24 Bit / 192 kHz'
Check it out at:
http://www.beis.de/Elektronik/ADDA24QS/AD24QS.html
My kit will be available in this month, and I am going to use it for my balanced(input) riaa preamp. As soon as I have it running I will lett you all know
Regards,
Frans.
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