2.5 ns is horrible
I believe that your e18 DAC might have adapted a synchronous master clocking mechanism so that you may take full advantage of the low jitter nature of signals output from your own high performance built-in USB-I2S interface board.
Otherwise, By using a conventional asynchronous fixed-frequency master clock of 80 or 100 MHz which brings unavoidable and inherent clock-tick derived jitter of 10 ns, your precious low jitter I2S signals will be degradated severely in the DAC chip.
One more my opinion. Direct signals originated from the inside of XMOS do not necessarily mean the I2S signals output from the USB interface board. Some skillful designer may use a direct or divided signal forked from an audio frequency dedicated external oscillator of low phase noise for LRCLK, BCLK or MCLK in I2S set.
Bunpei, are you saying that the ES9018 patented internal jitter-elimination processing is causing 10ns jitter?
Are you sure that you can draw parallels between the careless general purpose fractional division that takes place inside XMOS, and mechanism inside ES9018 that is actually designed to fight jitter?
Can you provide some evidence for this claim?
Yes, of course you can generate jitter and then you can deploy measures to cancel it. This is what SP/DIF is all about. My preference is to avoid it at first place.
Do you know who is doing jitter cancelling on xmos-based asynchronous USB to I2S interface? Is it done well? Are there any measurements, and are they backed by blind listening tests?
As you know, I've published measurements, and there is no hint for 10 ns jitter. The lack of jitter is backed by the testimonials of ALL of our users. They say that exaU2I is in a different league compared to the rest.
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... are you saying that the ES9018 patented internal jitter-elimination processing is causing 10ns jitter?
No. I think their jitter-reduction processing is a kind of stopgap measure of correcting sound intensity values by an interpolation. Route origin is in the clock tick gap between fs domain and DSP domain.
By the way, would you tell me the number of the patent that you indicate?
I'm not sure. However, ES9018 seems to have a computer chip-like architecture ( which point I favor much) for me. All events in the DSP-side domain within the chip might be under the control of system clock and they are aligned to a system clock tick.
Yes. However, I don't want to give any adverse effect to this thread of your business. Shall we have further technical discussions on ESS Sabre32 thread?
I agree with you. Your approach is correct, I think. Your exaU2I can perform very well with such DAC chip, DSD1794, WM8741, AK4399, etc. by employing a conventional synchronous master clocking scheme.
I have no facts on this point. However, I have a prospect of high confidence.
I avoid using the term "jitter" because the term is very ambiguous. If you show a phase noise chart instead, that will make me feel very comfortable.
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The exaSound e18 DXD DAC now supports OS X
The exaSound e18 DXD DAC now supports OS X. We offer two Mac drivers optimized for stereo and multichannel experiences at sampling rates up to 384kHz/32bit. Please check our website - www.exaSound.com.
The exaSound e18 DXD DAC now supports OS X. We offer two Mac drivers optimized for stereo and multichannel experiences at sampling rates up to 384kHz/32bit. Please check our website - www.exaSound.com.
I2S/Spdif input?
Hi,
the exaU21 and the E18 are output-only devices(?) I have a setup where I use the PC as a digital cross-over for various sources (tv, squeezebox, popcorn hour). Sometimes I even use the pc as the source, but in general I like to keep the screen off when listening to music. I switch from source to source via an external switchbox with a spdif-transformer. I wonder if you have given any thought about including some sort of input(I2s?) to the EXA? It would be great to know that the same bit-perfect drivers was handling the complete digital signal-chain to/from ASIO. I know that this requires some sort of synchronization (reclocking for instance) of the incoming data stream.
regards,
Øyvin Eikeland
Hi,
the exaU21 and the E18 are output-only devices(?) I have a setup where I use the PC as a digital cross-over for various sources (tv, squeezebox, popcorn hour). Sometimes I even use the pc as the source, but in general I like to keep the screen off when listening to music. I switch from source to source via an external switchbox with a spdif-transformer. I wonder if you have given any thought about including some sort of input(I2s?) to the EXA? It would be great to know that the same bit-perfect drivers was handling the complete digital signal-chain to/from ASIO. I know that this requires some sort of synchronization (reclocking for instance) of the incoming data stream.
regards,
Øyvin Eikeland
Hi Øyvin, both exaU2I and e18 are output-only devices. Your request is on the wish list.
My situation is similar to oyvine - for my DSP rig I currently use external sources routed into a SRC2496 that is slaved to the worclock output of my audio interface (Steinberg MR816x). this isn't the last word in convenience or ergonomics, but does provide flexibility.
I'm very interested in the E18, but am struggling to feel confident that the 'output only' is going to work in my setup. In particular, we use whole-house sync of our 3 squeezeboxes to play our library as well as streamed XM content, and I'm not entirely sure that I can replicate that even with JRMC.
IMHO a wordclock output would be a valuable addition and would probably be easier than a full audio input, but that assumes that you're using an external SRC and whatever signal chain you're using can use different interfaces for input and output - my current setup using Reaper probably can, but I'm not sure about things like JRMC etc.
I'm very interested in the E18, but am struggling to feel confident that the 'output only' is going to work in my setup. In particular, we use whole-house sync of our 3 squeezeboxes to play our library as well as streamed XM content, and I'm not entirely sure that I can replicate that even with JRMC.
IMHO a wordclock output would be a valuable addition and would probably be easier than a full audio input, but that assumes that you're using an external SRC and whatever signal chain you're using can use different interfaces for input and output - my current setup using Reaper probably can, but I'm not sure about things like JRMC etc.
Many customers have expressed concerns about the exaU2I shipping cost. We've just introduced a low-cost regular mail service to most overseas destinations. The regular mail small-package service is much cheaper, but unfortunately it is slower. There is no insurance and the shipments cannot be traced.
I'm shure this has been asked before somewhere, but cant find it ... so:
According to diy-guide, i should connect power to exaU2I from DACs DVCC. I use ackodac, that has split DVCC in top ring, and bottum ring: DVCC_T, DVCC_B
- which one to use?
- or just use a separate (superior) supply? (as ackodac will run syncron-mode with clock from exa)
According to diy-guide, i should connect power to exaU2I from DACs DVCC. I use ackodac, that has split DVCC in top ring, and bottum ring: DVCC_T, DVCC_B
- which one to use?
- or just use a separate (superior) supply? (as ackodac will run syncron-mode with clock from exa)
Yes, It is under development
When you say its "under development", do you mean that you are currently working with the JRiver guys on DSD streaming to the exaU2I and the e18 DAC?
Thanks,
Jim
I would use DVCC_T, it powers the top pad of the chip and it is located next to the data pins. It has to be 3.3v and it is important NOT to use a separate power supply.
The idea of using DVCC is to drive current trough the GMRs, into the DAC I2S inputs. exaU2I is isolated, so our power supply (USB) cannot do that. Additional power supply won't work either, you need a closed circuit for the current to flow. This job really belongs to the power source used for DVCC.
The synchronous mode of operation is not related to the DVCC connection. You need to do follow the same rules.
Regards,
George
Not important at all. exaU2I is optimized for audiophile grade playback, not recording. The requirements for recording are in conflict with the requirements for playback.
Artists need low latency to avoid echo while recording. Audiophiles need asynchronous operation to achieve immunity against computer hiccups. ASIO is a very powerful tool, and it can be used for more than one application.
Sorry, recording is out of topic here.
Artists need low latency to avoid echo while recording. Audiophiles need asynchronous operation to achieve immunity against computer hiccups. ASIO is a very powerful tool, and it can be used for more than one application.
Sorry, recording is out of topic here.
In 8 channel / 384 kHz mote the latency is around 7-11 ms. In 2 channel / 44.1 kHz the latency is 37 ms. Using the driver in 8 channel mode will not cause visible difference between video and audio timing.
I hope this helps.
what about 8/44khz?
it seem to have low latency good for audiophile playback.. but not for the rest...
for games, movies and music production .. near 2-6ms is the acceptable latency for the source..
it seems only RME have found how to do really low latency usb asio drivers..
Also, processing power taken by the asio driver should be extremely low.
RME are running their internal mixers on dsp chips..
it seem to have low latency good for audiophile playback.. but not for the rest...
for games, movies and music production .. near 2-6ms is the acceptable latency for the source..
it seems only RME have found how to do really low latency usb asio drivers..
Also, processing power taken by the asio driver should be extremely low.
RME are running their internal mixers on dsp chips..
exaU2U sounds much better than RME. When you optimize for one thing, you loose the other. Our ASIO driver is also very efficient, I can listen to 384 kHz on an Atom net-book.
DSP and bit-perfect are two incompatible consents. with exaU2I we offer bit-perfect.
good i think RME soso SQ is because of the dacs they use and not the usb to dac implementation.. they offer also bitperfect playback but i guess when enable it bypass the internal dsp mixer..
as for processing power, i guess the only way would be to compare them side by side on the same machine.
Your product seem interesting.. im trying to find a way to have a low latency active system (FIR crossover and DRC) with audiophile SQ(dac, usb to i2s) without have to buy the Deqx HDP-4
it's a mess.. it's seems there is no low cost alternative.. sad
as for processing power, i guess the only way would be to compare them side by side on the same machine.
Your product seem interesting.. im trying to find a way to have a low latency active system (FIR crossover and DRC) with audiophile SQ(dac, usb to i2s) without have to buy the Deqx HDP-4
it's a mess.. it's seems there is no low cost alternative.. sad
It would be nice if you had a mixer panel with a feature that lets you speed up or slow down the audio so you had perfect lip-sync. I know some of the player apps have this feature, but not all of them do. If you use Windows Media Center for watching DVDs and Blu-rays, you can't make these adjustments.