RME ADI-2 DAC Thread

[maszynista1987]

Really wanting to try stellia and zmf verite closed.

ZMF are great with Rme

 

[thiy71]

Excuse me,does anyone compare the performance between hugo1 and adi-2 to drive sennheiser hd650?
I want to know whether adi-2's amp is enough to drive hd650?
Thanks

 

[gimmeheadroom]

I do not like the RME amp with my 600s or 800s. The RME sounds good to me over my Fostex TRs and LCD-2C. Still not sure how I feel about it with the LCD-3F.

I don't have the Hugo. If you have Sennheisers I would recommend running the RME into a good balanced amp. Sennheisers can sound lifeless without enough power even though they get loud enough.

That said, I think it would be a terrible mistake not to buy the RME just because it doesn't have enough amp. To me it's a DAC, and it's superb at that. For one thing, it makes it enjoyable to use headphones you might not be able to listen to at all without the EQ. It's just full of great stuff. If I can run some headphones happily off of it that's just icing on the cake.

 

[darmccombs]

I turned it on, it was just missing something out of the quarter inch.. I plugged them in a phonitor (phonitor was powered by the rme adi-2 dac) and it drove it better, stronger more dynamic, to me a clear winner as far as power and clean goes, superior to the rme adi-2 dac Amp side (quarter inch) for iem I prefer rme.
I love the rme dac, sound quality DSP, vast features and pleasant screen. It was no slouch but I would say the phonitor, strickly looking at sound quality out of the quarter inch out was about 30/40 percent better than the rme... I'm a big fan, but I have to give credit where credit is due.
High power setting was turned on. Listened in SD sharp and slow, after initial thoughts on eq off I did eq on just to fool around and the rme did a great job as usual and the headphones responded well to minor eq changes as well.
Loved it all, but the clears were bright, the ethers not so much, going to skip utopia out of the rme, and just get the stellia.

Remember, amp is not solely about just driving something to Loud volumes. A 50$ amp can do all that.... But it is the *authority * or energy it can drive it with, more of an effortless sound, on caffeine... Ect.

Sure. Whatever makes you pet the rme more and give it the attaboy chat.

I listened for a few hours, to my best trial of volume matching by ear. I apologize for my 3/5db difference margin. The phonitor was significantly *better sounding * out of thr quarter inch. I haven't gotten the chance to try its balanced way of things but I would think there would be further changes.
The cans sounded punchier, crisper, cleaner vs the rme (quarter inch) in your beloved high power mode.

With that said is the phonitor worth 2k for it to make it happen? Not to me....
Would I buy one used at 1.2k? Idk...
Does it have the incredible price performance the rme adi-2 dac has.... Pfffft No!
Did it sound better to me with every can I put in it? Yes....
Significantly?... I guess.

Are you butt hurt?.... Possibly

Thanks for the feedback. I understand exactly what you are seeing. I’ve seen this in some home systems a few times.

 

[SupperTime]

Thanks for the feedback. I understand exactly what you are seeing. I’ve seen this in some home systems a few times.

I hope you come to enjoy it more. If not. It's really a whole lot cheaper to keep mojo, return rme adi-2 and just buy a jds atom or the equivalent for a strong capable amp dac combo, nothing wrong with that!

 

[darmccombs]

I hope you come to enjoy it more. If not. It's really a whole lot cheaper to keep mojo, return rme adi-2 and just buy a jds atom or the equivalent for a strong capable amp dac combo, nothing wrong with that!

Oh, I am keeping the RME. It goes great in my home system used as a DAC only. I really like the unit. If I can get the RME working better than the mojo with my headphones, I may even buy a 2nd RME.

I will be doing more testing with the RME DSP settings to see how good I can get my Z7m2 HP to sound. Right now that is my focus.

 

[pororom]

I ask for help

I have the ADI 2 Dac and I only used it as a headphone amplifier, but now I want to connect 2 Genelec

How should I do it?

Do I have to put something between the monitors and the ADI or from the ADI can I adjust the volume?

I am concerned about the configuration of the ADI output volts and the Genelec input volts. I have no idea how to set it up

I don't want to break the devices

 

[gimmeheadroom]

The Genelecs have volume control or you prefer to control the volume from the RME?

You can do it either way. Either you run the RME at full output and adjust volume on the speakers, or you turn the RME volume down to something like -60 dB, turn the Genelecs on full power, and then adjust the volume from the RME.

 

[pororom]

The Genelecs have volume control or you prefer to control the volume from the RME?

You can do it either way. Either you run the RME at full output and adjust volume on the speakers, or you turn the RME volume down to something like -60 dB, turn the Genelecs on full power, and then adjust the volume from the RME.

I don't think the Genelecs have volume control

https://www.genelec.com/8030c

I want to control the volume from the ADI

Do I connect the XLR cables and always turn on the speakers first?

 

[gimmeheadroom]

I don't think the Genelecs have volume control

https://www.genelec.com/8030c

I want to control the volume from the ADI

Do I connect the XLR cables and always turn on the speakers first?

You can leave the speakers on and when music is playing from the RME they will wake up.

 

[CaptainFantastic]

So now that it has been a few months since the release of the ADI-2 v2 with the AKM4493, has anyone had a chance to really test the old and new models for perceptible sound differences?

For example articles like this (yes, it is marketing) suggest there is a substantial improvement in the technology between AKM4490 and AKM4493... the whole part about "Velvet sound technology", "improved acoustic sound expression", "lower distortion", etc. Does any of this translate into a better sound experience between ADI-2 v1 and v2?

https://www.marketwatch.com/press-r...-best-selling-premium-high-end-dac-2017-12-08

Thanks.

 

[gimmeheadroom]

After RME themselves said there is no audible difference it seems folly to spend more time on this the noise floor is already well below human audibility.

 

[SupperTime]

So now that it has been a few months since the release of the ADI-2 v2 with the AKM4493, has anyone had a chance to really test the old and new models for perceptible sound differences?

For example articles like this (yes, it is marketing) suggest there is a substantial improvement in the technology between AKM4490 and AKM4493... the whole part about "Velvet sound technology", "improved acoustic sound expression", "lower distortion", etc. Does any of this translate into a better sound experience between ADI-2 v1 and v2?

https://www.marketwatch.com/press-r...-best-selling-premium-high-end-dac-2017-12-08

Thanks.

I had both and now have v2. But only a few days with v1.
There is an additional filter now and with that on it does sound marginally different, rme said no other changes. And I too haven't noticed anything, but again, I didn't have much time with the gen 1. About 6 hrs

 

[MLGrado]

Stream of Consciousness Day 1

• Speaking of DSD... you can do direct DSD via line outs only since you're not allowed to manipulate the volume. It silences the headphone outs if you turn on DSD Direct. I believe it converts DSD to PCM internally if you leave it off, which is default.

The AK chip does not require dropping to pure PCM as we think of it to perform DSD volume control. DSD input into the chip goes to a separate filter vs. PCM when not using the direct bypass feature. (Direct bypass can ONLY be used on line out.

If you input DSD and aren't using direct bypass, there is a volume stage in the 'DSD Domain'. There are several technical documents out there that somewhat explain the process. Sample rate is unchanged, but the result is a file with multiple bits per sample. One is lead to think these volume controls do not filter/decimate the signal. But more commonly I believe filtering is done, and it should take advantage of the high sample rate to implement a very slow, smooth filter that retains maximum information and impulse response. Then this 'intermediate signal' in the 'DSD Domain' is converted via the Delta Sigma Modulators on the chip and the accompanying final stage filtering.

Now, you can throw all of that out if you use any DSP other than volume when listening to DSD with Phones or in 'non-direct' mode via line out. The DSD signal must be converted to standard PCM to implement the multiplicity of RME DSP features. This would happen pre AKM chip. All data would go into the chip as PCM, never being routed to the on-chip DSD filter.

Of course RME could be avoiding completely the onboard DSD stage, never using the onboard DSD filter at all, Sending DSD via 'standard' PCM to the DAC. The exception being when line-out direct mode is used, and you get 'pure' DSD conversion with no intermediate DSP.

That's the best I have figured out after many hours of painstaking research on the subject. Other DSP than volume could be done in the so called 'DSD Domain', but that requires very complex systems probably way beyond the scope of this product. For EQ, Crossfeed, and similar DSP's, PCM remains the best way, if not the ONLY way to get certain things done.


By the way, I can tell you the Direct DSD via line out in marvelous on this DAC. If you want a 'true DSD' experience, this is one of the few products that will get you there. Most others involve intermediate DSP.

 

[Triodemode]

The AK chip does not require dropping to pure PCM as we think of it to perform DSD volume control. DSD input into the chip goes to a separate filter vs. PCM when not using the direct bypass feature. (Direct bypass can ONLY be used on line out.

If you input DSD and aren't using direct bypass, there is a volume stage in the 'DSD Domain'. There are several technical documents out there that somewhat explain the process. Sample rate is unchanged, but the result is a file with multiple bits per sample. One is lead to think these volume controls do not filter/decimate the signal. But more commonly I believe filtering is done, and it should take advantage of the high sample rate to implement a very slow, smooth filter that retains maximum information and impulse response. Then this 'intermediate signal' in the 'DSD Domain' is converted via the Delta Sigma Modulators on the chip and the accompanying final stage filtering.

Now, you can throw all of that out if you use any DSP other than volume when listening to DSD with Phones or in 'non-direct' mode via line out. The DSD signal must be converted to standard PCM to implement the multiplicity of RME DSP features. This would happen pre AKM chip. All data would go into the chip as PCM, never being routed to the on-chip DSD filter.

Of course RME could be avoiding completely the onboard DSD stage, never using the onboard DSD filter at all, Sending DSD via 'standard' PCM to the DAC. The exception being when line-out direct mode is used, and you get 'pure' DSD conversion with no intermediate DSP.

That's the best I have figured out after many hours of painstaking research on the subject. Other DSP than volume could be done in the so called 'DSD Domain', but that requires very complex systems probably way beyond the scope of this product. For EQ, Crossfeed, and similar DSP's, PCM remains the best way, if not the ONLY way to get certain things done.


By the way, I can tell you the Direct DSD via line out in marvelous on this DAC. If you want a 'true DSD' experience, this is one of the few products that will get you there. Most others involve intermediate DSP.

Nice summation on how DSD gets adulterated in most DACs, including those which utilize ESS chips as I have read. AK and Texas instruments are two of only a few DAC chips that allow DSD without any bit modification.