New Lavry DA11 DAC

[Sherwood]

haha, "remote ready". Pretty much anything with IR is "remote ready" with learning remotes.

[Voltron]

Pfft, you just want to win another won.

Third time is the charm!

[Voltron]

haha, "remote ready". Pretty much anything with IR is "remote ready" with learning remotes.

DA10 doesn't have IR, so this is an improvement in my book.

[aardvark baguette]

They pulled a Benchmark. The price is $1480.

[Dreadhead]

But judging from the controls on the front, there's no way to select a completely different transfer function. It seems a lot simpler than what I'm contemplating (which can't be done in analog... the only DSP that can be done in analog is somewhat gimmicky in the sense that it's at best a first-order approximation of reality).

DSP certainly makes it a lot easier but what you said was "DSP-based convolution of the signal based on an average head or room transfer function" which is likely what it is.

Well I've actually written my own convolution generator etc for DSP and I see nothing that can't be done in analog (admittedly not with the control you get in DSP). I'm getting an SPL phonitor for review (bloody shipping is taking forever) which includes a very complex crossfeed and I'll know better how they do working in the analog domain.

[Filburt]

Wasn't impressed by the DA-10; maybe this is an improvement.

Anyway, total conjecture time...

I see no mention of CrystalLock, and it specifies 200KHz sample rate max. It also mentions digital volume control, but says it's "analog"; not sure if this means there's some sort of digitally-controlled resistor ladder attenuator system or if it's just marketing-speak for something on-chip with the d/a. Anyway, the other two features might suggest: a. switch to just using an ASRC or something involving one, and b. the 200KHz sample rate may mean change in D/A converter, since the AD1955 (used in the DA-10) at least doesn't spec 200KHz. If so, this could suggest a switch to the ESS Sabre (or Sabre32), as 200KHz is its max spec'd sample rate, and it contains an ASRC-based jitter rejection component. I think the Sabre also has digital volume control although I don't recall off-hand. If it's using a Sabre, I wonder if it's using the voltage outputs or the current outputs.

[Filburt]

Want to re-emphasize, I have no idea what chip it uses; mostly I just wondered some others' thoughts on the matter of differences from the DA-10 spec. I can't remember if I ever tried feeding the AD1955 eval board a 200KHz sample rate or not; maybe it can accept it.

Perhaps Mr. Lavry will at some point say what it uses. One thing that does leave me rather skeptical about the proposition is I guess I would have expected it to have been part of the announcement.

[grawk]

Zach should buy it, crack it open, and post very high resolution pictures of it

[AlanY]

The DA10 had the same spec. From the DA10 manual:

"Use WIDE for non-standard frequencies between 30KHz and 200KHz."

[Filburt]

The DA10 had the same spec. From the DA10 manual:

"Use WIDE for non-standard frequencies between 30KHz and 200KHz."

Ah, OK. I didn't see that in the manual. Thanks for pointing that out

So, are there any salient differences you're noticing between this and the DA-10 apart from USB input?

[slwiser]

The wide uses ASRC like the benchmark. The narrow and crystal uses SCR. The DA10 has two DAC chips inside it for different settings, the AD1955 (integer multiple) and AD1896 (non-integer multiple). It up-samples everything. Just that when going ASCR it does the non-integer upsampling and when doing SCR it does interger upsampling.

[Filburt]

The wide uses ASRC like the benchmark. The narrow and crystal uses SCR. The DA10 has two DAC chips inside it for different settings, the AD1955 (integer multiple) and AD1896 (non-integer multiple). It up-samples everything. Just that when going ASCR it does the non-integer upsampling and when doing SCR it does interger upsampling.

The AD1896 is the ASRC. It is not a D/A converter. I don't understand what you're trying to say with the integer/non-integer multiple unless what you mean is the ratio between original SR and the resampled SR is not an integer, however it still doesn't make much sense since the AD1955 doesn't resample. There was a thread on DIYHifi.org a while back about the AD1896 and CrystalLock mode - DIYHiFi.org; View topic - Lavry DA10 DAC & CrystalLock?. I haven't seen much discussion about the matter since then; kind of wonder what came of it all.

[AlanY]

I haven't seen much discussion about the matter since then; kind of wonder what came of it all.

I think the verdict was that later revisions of the DA10 firmware always use the AD1896 and never use CrystalLock/Narrow modes regardless of how the switches are set. Though Lavry changed the manual to include something about a 50 second delay on mode switching, which might mean that the critics were measuring wrong (i.e. they were switching too frequently), but I kind of find that doubtful. Also, it looks like there's no mention of CrystalLock/Narrow in the DA11, and no interface for selecting them, which would seem to confirm that Lavry has settled on just using the AD1896 all the time. Not a bad choice overall in terms of sound quality.

[slwiser]

The AD1896 is the ASRC. It is not a D/A converter. I don't understand what you're trying to say with the integer/non-integer multiple unless what you mean is the ratio between original SR and the resampled SR is not an integer, however it still doesn't make much sense since the AD1955 doesn't resample. There was a thread on DIYHifi.org a while back about the AD1896 and CrystalLock mode - DIYHiFi.org; View topic - Lavry DA10 DAC & CrystalLock?. I haven't seen much discussion about the matter since then; kind of wonder what came of it all.

So I am getting a bit mixed up here maybe, but I was attempting to refer to what Dan Lavry posted in one of his threads on his forum and I have captured it in a pdf. I copied Dan's words below.

My take from reading this is that the AD1896 is doing the work of the ASRC while it would appear that the AD1955 does SCR without having to use the ASRC (non-integer) feed from the AD1896. At least that is what I am getting from reading this. The spec of the AD1955 shows that it upsamples using SCR (integer). If anyone is interested I can email the pdf of this thread.

For myself this is the best single explanation of what Dan Lavry has asserted in what he has done with the DA10 using the Crystal, Narrow and Wide settings. This post was a few months after the initial changes that have been noted were done I think. The DA10 was first introduced around Jan. 2006. I have had one since Feb. 2006 and got my second one last year. They sound the same to me.

Posted Sept 6, 2006 and the text in bold and underlined is my highlighting.

"The DA itself is always operating in an up sampled mode which is synchronous. This is needed because it allows a real world design of an anti imaging filter. Prior to the concept of up sampling, the analog anti imaging filters were made of a lot of parts, but never enough to yield a reasonable performance. It would take dozens of opamps and precision resistors and caps to have the proper filtering, and while doing so, other things fall apart...Also, there is another reason for up sampling:

a DA with no up sampling has very non flat amplitude vs. frequency response. In theory, a DA is perfect, because the samples are

[Filburt]

OK, I think I understand the confusion now.

What the AD1955 has is an oversampling filter, which in very simplified terms has the net effect of pushing imaging products away from fs/2 (22.050KHz in the case of CD Audio), so you don't need a steep filter there which is both costly as well as is difficult to do well. You could do a sinc comp at the output to combat the rolloff, but oversampling I think is the better solution. Anyway, not sure why he's referring to the oversampling filter as an "upsampling" filter, as people usually associate that with SRC, but I guess it's inconsequential. I don't know what you mean by SCR, though; not familiar with that term, however the filter in the 1955 does use interpolation, and I guess it's "synchronous" and "integer" in the sense that a. it isn't a sample rate converter (so it's "synchronous" with the original Fs) and b. the oversampling frequency is an integer multiple of the sample rate.

Anyway, aside from that, what I had asked about above (and what Alan answered) pertained to whether the ASRC is actually disabled in CrystalLock mode.

[AlanY]

The manual for the DA11 is now posted on the Lavry site. No mention at all of CrystalLock so they've almost certainly abandoned the steerable clock concept, as they probably also did in later revisions of the DA10.

Looks like it uses Philips TV remote codes, so you don't have to purchase a separate remote. They get positive points from me for this. I can't believe Bryston charges an extra $500 for the remote (not the remote circuit, just the remote handset) for their new BDA-1 DAC.

[slwiser]

OK, I think I understand the confusion now.

What the AD1955 has is an oversampling filter, which in very simplified terms has the net effect of pushing imaging products away from fs/2 (22.050KHz in the case of CD Audio), so you don't need a steep filter there which is both costly as well as is difficult to do well. You could do a sinc comp at the output to combat the rolloff, but oversampling I think is the better solution. Anyway, not sure why he's referring to the oversampling filter as an "upsampling" filter, as people usually associate that with SRC, but I guess it's inconsequential. I don't know what you mean by SCR, though; not familiar with that term, however the filter in the 1955 does use interpolation, and I guess it's "synchronous" and "integer" in the sense that a. it isn't a sample rate converter (so it's "synchronous" with the original Fs) and b. the oversampling frequency is an integer multiple of the sample rate.

Anyway, aside from that, what I had asked about above (and what Alan answered) pertained to whether the ASRC is actually disabled in CrystalLock mode.

It would seem that the ASRC chip could still be active and could be doing both non-integer and integer oversampling when needed. So it would appear to be active all the time. It could be that for a signal set to a common frequency that the ASRC chip is set to do integer multiples and only during the Wide would it do non-integer oversampling. This is conjecture on my part.

[Filburt]

It would seem that the ASRC chip could still be active and could be doing both non-integer and integer oversampling when needed. So it would appear to be active all the time. It could be that for a signal set to a common frequency that the ASRC chip is set to do integer multiples and only during the Wide would it do non-integer oversampling. This is conjecture on my part.

The AD1896 is a sample rate converter. It is there to convert sample rate, and an incidental effect of this process is it can reclock the signal and thereby provide jitter rejection. Although, I've seen it suggested that this is a poor use of an ASRC because the jitter will be integrated into a sampling error post-SRC. I'm not sure if there is a verdict on how bad this problem is in real-world application.

Aside from that, I'm not exactly sure what you're trying to say. It sounds like you are trying to talk about ratios (e.g. integer/non-integer) but mixing the oversampling operation of the AD1955's filter with the resampling operation of the AD1896. Both use interpolation, filtering, and so forth but the AD1896's function is distinct from that of the AD1955.

Anyway, it seemed to me that the main issue in the DIYHifi discussion was over whether CrystalLock was an independent, synchronous reclocking scheme that did not rely upon the use of an ASRC.

[slwiser]

Bottom line is that I can tell a distinct difference between all three settings but not as much between the Crystal and Narrow settings. If all did ASCR the same way by creating intermediate interpolations as opposed to adding zeros for oversampling, I would think this would not be as clear.

[dvse]

Bottom line is that I can tell a distinct difference between all three settings but not as much between the Crystal and Narrow settings. If all did ASCR the same way by creating intermediate interpolations as opposed to adding zeros for oversampling, I would think this would not be as clear.

This might have something to do with (otherwise non-functional) crystallock circuitry being switched off in the wide mode and affecting output volume. If I recall, it was speculated at diyhifi that Lavry disabled the crystallock in response to the frequent static problems in the early production units. His subsequent removal of any mention of synchronous reclocking from the manual goes a long way to confirm this.

[Dan Lavry]

The way I read it there shouldn't have been any difference.

It has always been my understanding that both optical and RCA type connections are S/PDIF but maybe I have that wrong? TOSLINK is an official trademark held by Toshiba so if the connection is the standard EIAJ optical connection it may not be in their interest to use that terminology at all.....

Hello,

I am up to my ears with work, but I could not ignore your post...

There are formats, and there are hardware.

AES format and SPDIF format are different, but there are also a lot of format similarities. The data (the bits that carry the music) is virtually the same in both formats.

The XLR, RCA and Optical connectors are very different. Typically the XLR carries "high power" (2-5V into 110 Ohms), the RCA is based on .4V into 75 Ohms, and the optical is a completely different signal (light, not current).

It is true that SPDIF started as a particular standard calling for RCA, and Toslink was started with a certain aim (consumer 44-48KHz). Similarly, the AES/EBU has some history.

But the "lines have been blurred". Some Toslink implementations can now handle 24bits at 96KHz, one can send and retrieve SPDIF music data on an XLR and so on... I am talking about the music data only, but for a DA, the music portion of the format is what one needs...

The DA11 can receive either spdif or AES format on any of the three connectors. That is very handy for driving multiple sources into the same DA. Clearly, I would choose XLR's for long cables, or for very electrically noisy environment - the XLR offers higher power plus balanced transformer coupling. Optical is pretty good, and RCA is somewhat weak, but works fine with short cables.

AD's require all sorts of parameters to be set by the user, such as the format type, number of bits, sample rate...

But with a DA, much of the operation can be done "automatically".

You can feed the DA11 as few or as many bits as you have (16-24 bits), and it will process them correctly. A 16 bits word is "viewed as" a 24 bits word with 8 zeros... You can feed the DA11 say 44.1KHz, or 48 or 88.2 or 96KHz and it will measure the actual sample rate and display it (in setup mode). It knows what is coming in; it knows what to do with it.

You can send to the DA11 either AES or SPDIF to the XLR, RCA and TOSLINK. You can send an AES signal into the optical, or a SPDIF into the XLR, or any other "combination". The DA11 will play the music present on the selected input.

Regards

Dan Lavry

Lavry Engineering - Unsurpassed Excellence

[Voltron]

Hello Dan, and welcome to Head-Case! I know you are busy and not active on the other headphone site, but we would love to get your input around here. It is a smaller group with a higher average understanding level, and knowledgeable members are free to speak their minds. You may notice a higher BS sensor as well, so you might need to be prepared to engage in a different way than you might do elsewhere.

Cheers, and I do hope to see you and Priscilla in Los Angeles for the big meet.

[Dan Lavry]

Hello Dan, and welcome to Head-Case! I know you are busy and not active on the other headphone site, but we would love to get your input around here. It is a smaller group with a higher average understanding level, and knowledgeable members are free to speak their minds. You may notice a higher BS sensor as well, so you might need to be prepared to engage in a different way than you might do elsewhere.

Cheers, and I do hope to see you and Priscilla in Los Angeles for the big meet.

Hi Voltron,

Thanks for your comments. I will look up the date for the big meet. The next few month are very busy for me, some of it is about meetings and conventions, but I will try to make it. It is not too far from Seattle...

Regards

Dan Lavry

[HeadphoneAddict]

I was impressed with Plaidplatypus's DA10, so I'll have to keep my eyes on the DA11. The only thing I'd want different is the spot where the XLR-in is located to instead be a pair of RCA-out, so it has both SE and balanced output and can drive two amps (my balanced dynamic rig and my SE Woo GES). When we tried plaid's DA10 in my rig, we connected the XLR out into my Square Wave and used the RCA on the Sq Wave as loop out to feed the GES, but a direct connection to the DAC would be preferred.

[Dan Lavry]

I was impressed with Plaidplatypus's DA10, so I'll have to keep my eyes on the DA11. The only thing I'd want different is the spot where the XLR-in is located to instead be a pair of RCA-out, so it has both SE and balanced output and can drive two amps (my balanced dynamic rig and my SE Woo GES). When we tried plaid's DA10 in my rig, we connected the XLR out into my Square Wave and used the RCA on the Sq Wave as loop out to feed the GES, but a direct connection to the DAC would be preferred.

The DA11 pakage includes a couple of good XLR female to RCA female adapters. I say "good" because the quality of such adapters ranges from poor to good. The specifications of the unit are met with the adapters in place. The DA11 lets you set unbalanced Pin 2 hot, unbalanced Pin 3 hot or balanced operation. The settings are controlled from the front pannel, so there is not need to open the chassis and set jumpers.

Other headphone related improvments (relative to DA10) are:

1. One can turn off the rear pannel signals, so the only active analog outout is the headphone. This is also controlled from the front panel.

2. There is a circuit that eliminates the "loud clicks" that take place during power up and power down (turnning the unit on and off). The DA10 has a power up "click protection", but not power down. The DA11 has both power up and power down protection.

The idea is to protect the ears. At very high volume, one can still hear a tiny power down click, but at much lower volume then the music. Without any protection, the click is huge. I am in the habit of removing the headphones prior to power down...

And of course, the >PiC< (Playback image control) is very much oriented to headphone. Almost all of the music material is

produced by monitoring through speakers, and the PiC enables the headphone listen to a stereo image closer to what was intended by the mixing and mastering people.

Regards

Dan Lavry