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Topic: exaU2I-Multichannel 384 kHz/32bit Asynchronous USB to I2S Interface (Read 16432 times) previous topic - next topic
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exaU2I-Multichannel 384 kHz/32bit Asynchronous USB to I2S Interface

Hello Everyone,

Please allow me to introduce a new experimental design for 8-channel 384 kHz / 32bit asynchronous USB to I2S interface. exaU2I was originally developed as a hobby project. We were looking for a hardware solution that can push to the limits the capabilities of Foobar for high-end playback.  At the present time the interface is available as a DIY kit. It clearly has the potential to rival the quality of traditional audiophile gear. The current design targets ES9018 DACs. It can be used with almost any device with I2S input.

Here are the details:
• Project homepage – http://www.exaDevices.com
• Sampling frequencies (kHz): 44.1, 48, 88.2, 96, 176.4, 192, 352.8, 384
• Resolution (bits): 16, 24, 32
• 8, 4 or 2 channels for sampling rates up to 192 kHz
• 4 or 2 channels for 352.8 (DXD) and 384 kHz
• Galvanic isolation between the USB ground and the I2S outputs.
• Two quartz oscillators for the 44.1 kHz and 48 kHz sampling rate groups
• FPGA implementation
• True asynchronous operation
• Sample rate LED indicators
• Four I2S data outputs (8 channels) powered by the DAC power supply (3.3V or 5V)
• Bit-perfect ASIO driver completely independent from the Windows sound system; No software volume control or mixing. Automatic sampling rate switching.
• Works on Windows XP, Windows Vista and Windows 7.
• Tested with Foobar and J. River Media Center


    Any feedback is highly appreciated.

exaU2I-Multichannel 384 kHz/32bit Asynchronous USB to I2S Interface

Reply #3 –
$430 for nothing more than a digital-digital bridge? That sounds absurdly expensive, unless I'm totally misunderstanding something. o.O

exaU2I-Multichannel 384 kHz/32bit Asynchronous USB to I2S Interface

Reply #4 –
Quote from: Canar on
$430 for nothing more than a digital-digital bridge? That sounds absurdly expensive, unless I'm totally misunderstanding something. o.O


Well, it is just a bridge - USB to I2S. It is developed in-house from scratch and it is hand built. exaU2I is a software / hardware solution that completely bypasses the Windows sound system and it also eliminates virtually all the noise and jitter caused by the computer. It is bit-perfect, 32bit, 384 kHz, 8 channels. Check the testimonials on our website - our users consider exaU2I to be in a league of its own in terms of sound quality.

exaU2I was built initially for our own use. We needed a piece of hardware that can unleash the full power of Foobar. An interface that is simple and designed to do one thing well - to stream the sound out of a computer with minimal interference. 

Canar please let me know if you are interested in testing the device.

Happy Holidays,
George

exaU2I-Multichannel 384 kHz/32bit Asynchronous USB to I2S Interface

Reply #5 –
Quote
to stream the sound out of a computer with minimal interference.


So you feed a USB signal and convert it to I2S to feed into what ? A Dac ?

I do not get the purpose of that item. Why would I use an outboard DAC when I have one on my sound card that does the same without the need of extra cabling?
Or I could feed the signal directly via spdif to an converting amp.

Audiophile idiocy again ?

exaU2I-Multichannel 384 kHz/32bit Asynchronous USB to I2S Interface

Reply #6 –
No, not this time. Actually the idea is to escape from the voodoo magic of modern audiophile myths. When I started developing this device my thinking was that there are too many layers in the chain of sound processing. It is hard to know what's going on. I wanted to remove any unneeded complexity and to get a clean signal path. My playback device of choice was a PC. The first quest was to find a bit-perfect player. It took me a while to appreciate Foobar.

The next obstacle was the Windows sound system. It is hard to understand, verify and control what is going on in there.  So I developed my own ASIO driver. It is independent from the Windows kernel and to me it was a step forward - I knew for sure that there is no mixing,  re-sampling and volume control. I got the luxury of automatic sampling rate switching.  I like to play the source material at the native sampling rate, bit-perfect.  Nothing added, nothing taken away.

Then there is the influence of the computer hardware. Computers generate high-frequency noise. Their power supplies are not designed for audio purposes. Windows clocks are not precise and music playback is randomly interrupted by high-priority tasks. The result is lots of noise and jitter. Internal computer sound-cards must deal with these problems. It is actually easier to  avoid all this noise and jitter by using an external DAC.

Following the sound streaming chain, the next challenge is to take the signal out of the computer. Consumer legacy interfaces like SPDIF cause jitter of their own. USB Audio 2.0 can do the job, but is it really complicated. I needed a simple a verifiable solution. I  also wanted to experience  32bit detail at high sampling rates and more than two channels. Foobar's internal processing is 32bit, why compromise with the interface? Eventually I had to build the missing DAC-Computer link.

Finally, to experience realistic, detailed and dynamic sound you need a good DAC. Internal sound cards are multi-purpose devices. A good DAC needs a good master clock, and a quiet power supply. It needs isolation from the computer noise and ground loops issues. The DAC must offer low distortion and superb signal to noise ratio. I've also built a complete DAC using the ES9018 chip.

exaU2I-Multichannel 384 kHz/32bit Asynchronous USB to I2S Interface

Reply #7 –
Worthy goals, maybe.

I run m-audio souncard on a DIY computer that uses a i7 processor, and when foobar is running less than 5% of the processor is used.
The signal passes from foobar accessing the hardrive, processing the signal to 48kHz and sending it via spdif to a DAC build into my amp that decodes 48kHz to send to the amp.
A fairly direct path with ONE cable connection (besides the sata from the harddrive).

I had previous to that a fairly costly audio setup, including a pioneer 59AVI multi format player that replaced a Micromega stage 2 cd player that replaced a Denon 1520 Cd player, with 3 bryston amps and a digital crossover (dcx2496) that replaced a marchand 48db/octave analogue crossover and a room correction processor deq 2496, all driving the bryston amps into a DIY sub/sat set through a Bryston preamp. Later the DIY speakers were replaced by kef's 104/2 and 2 DIY subs isobaric config.
I might say that I am not a stranger to decent sound. All steps to replace equipment (especially the replacement of the analogue x-over by the DCX and the incorporation of the DEQ) were done after careful listening sessions.

After processing all analogue and digital signals through a SRC by behringer, and feeding the processed signal further down the chain, replacing the Bryston preamp with a six channel passive  volume control I still was not able to experience any sound degradation - the opposite was true, because I finally had gotten rid of any noise from the speakers (produced by various analogue processors) on quiet passages at high volumes. The next step was rather logical - get rid of the whole shebang and go computer. I replaced the Brystons with Hypex AS amps and set up my own server.

I have compared asio to non asio signals and could not discern any problems or sound degradations.
I use asio because this gives me better control over the m-audio soundcard, so the only volume control that I can use is the one in foobar, the signal  goes through the software return path of the soundcard direct to the spdif output, bypassing all sound card controls and the window stack.
This permits me to set the master control of the soundcard at one level, and the foobar control at another one. This way I can switch from one source to another without sudden volume changes due to input level differences.
I also run an analogue signal from my phono preamp to foobar "record". In  foobar however it is not possible to select external inputs directly, this signal has the enter through windows path.

I was never able to make direct a/b comparisons between the analogue/digitized setup and the full digital setup, but I have some reference tracks (both digital and analogue) whose sound qualities I know quite intimately having used those for comparisons over at least 8 years and at least 100 times.
Because of my experiences I am rather doubtful of your claims.

exaU2I-Multichannel 384 kHz/32bit Asynchronous USB to I2S Interface

Reply #8 –
Unless you provide level-matched blind listening tests you won't get the desired responses from this forum. The testimonials on your side are anecdotical evidence only.

exaU2I-Multichannel 384 kHz/32bit Asynchronous USB to I2S Interface

Reply #9 –
Quote from: exa065 on
Then there is the influence of the computer hardware. Computers generate high-frequency noise. Their power supplies are not designed for audio purposes. Windows clocks are not precise and music playback is randomly interrupted by high-priority tasks. The result is lots of noise and jitter. Internal computer sound-cards must deal with these problems. It is actually easier to  avoid all this noise and jitter by using an external DAC.

...the question being can anyone actually hear this noise with a decent (or even onboard) sound card? Very unlikely.

Quote
Finally, to experience realistic, detailed and dynamic sound you need a good DAC. Internal sound cards are multi-purpose devices. A good DAC needs a good master clock, and a quiet power supply. It needs isolation from the computer noise and ground loops issues. The DAC must offer low distortion and superb signal to noise ratio. I've also built a complete DAC using the ES9018 chip.

No I don't, the DAC on my PC works just fine thankyou.

Is it me or is this just audiophile fluff? Unless you can provide some ABX tests running this against some other hardware you'll get nowhere on here.

exaU2I-Multichannel 384 kHz/32bit Asynchronous USB to I2S Interface

Reply #10 –
Quote from: onkl on
Unless you provide level-matched blind listening tests you won't get the desired responses from this forum. The testimonials on your side are anecdotical evidence only.


You are right, the only way to know is to do level-matched, blind listening tests. The best thing is to have the testing done with the participation of an independent third party. Perhaps there are well respected forum members that  live in the Greater Toronto Area, where I am based.  It would be fun to do the testing together.