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RME User Forum → ADI-2 & 2/4 Pro series, ADI-2 DAC series → auto stand-by ADI-2 Pro
The Auto Stand-By setting in my ADI-2 Pro F BE R is not working regardless of the time selected. It did it for a while after the FM update that expanded the time choices but now it doesn't. Thanks
It only works if no signal is present.
There was an issue with the analog input that is fixed or not in the latest FW.
Thank you, Kais. I understand it is not a timer to go into standby after a certain number of minutes of operation (signal or not). It is a control to turn the unit into standby a certain amount of minutes after the signal has stopped. I had the unit set to go into standby 30 minutes after the signal had stopped. It worked for a while but now it does not work at all, at any time setting.
Sorry but I don't understand the second part of your comment. I have the latest firmware. Is this something RME is aware of?
dr.larkos
https://forum.rme-audio.de/viewtopic.php?id=35351
In the current firmware 115 (not yet released for Windows) the analog inputs also prevent standby mode. Trigger threshold is -70 dBFS, so AutoStandby should still work for you.
Thank you fieldstu. I understand now. I remember asking MC a few years back about the normality of that residual low-level signal present at the Pro's output meters when using the analog input, even though there is actually no signal coming from the source. I guess in my case the residual signal is above the threshold so it keeps the unit going on, preventing auto standby. I will investigate and confirm if the problem is only when using analog input.
dr.larkos
I remember asking MC a few years back about the normality of that residual low-level signal present at the Pro's output meters when using the analog input, even though there is actually no signal coming from the source. I guess in my case the residual signal is above the threshold so it keeps the unit going on, preventing auto standby.
If you don’t need the analog input, setting it to +24 dBu Reference Level might help reducing this signal enough to not trigger the timer.
And / or use dummy plugs with shorted XLR pins 1+2+3 to calm down the inputs.
And maybe check if the incoming signal disappears if you disconnect the input cables (assuming unbalanced)
Kais and fieldstu, thank you, guys! All makes sense. I do use the analog input to play vinyl in order to benefit from the room equalization that I have loaded in the ADI-2 Pro...Incidentally, when doing this the analog to digital conversion has to be using PCM, not DSD (playback of DSD files cannot benefit from EQ and many of the ADI-2 Pro's many DSP features and functions), correct?
Thanks again and cheers.......................dr.larkos
Correct, that’s the limitation of the DSD format:
NOTHING can be done to the signal in it’s native format, no EQ or any DSP processing, not even volume control.
Therefore, as end user delivery format it’s a fake in almost every case, as no studio can work with it except for simple transfer from analog to digital.
Every source for DSD was either analog (tape e.g.) or PCM before.
A pure DSD recording, front to end, would only be possible if done like former “Direct To Disc”-recordings:
Musicians playing live, ad hoc being mixed and recorded to an final, un-changeable result.
That’s not how music was or is produced since tape exists and evolved to DAW, with very few exceptions.
The idea behind DSD was, getting an end user format with a higher frequency resolution than CD without extending the data rate too much, for delivery on DVD format discs.
Even getting rid of the reconstruction filter, the real audible limitation of the 44.1 kHz CD format.
DSD can store very high frequencies at rel. low bitrates.
The trick is, high amplitudes of high frequencies are not possible (only one bit amplitude change per sample).
This is no practical limitation, as those don’t exist in real world audio recordings.
informative & helpful post above
informative & helpful post above
Also look at the file sizes, too high storage requirements for really no reason.
The biggest bummer is that all productions base on PCM, as Kai says.
Only by converting PCM to DSD format, you do not get an increase in audio quality.
Similar with up sampling from 44.1 to 192 or even higher sample rates.
It would have been sufficient to simply change from single to double speed (88.2 kHz).
Then all AD/DA filter issues would have been moved to the inaudible area over 20 kHz and voilà a real solution.
The only problem: you won't get as much music material to a CD.
But in this case, why not simply switch to DVD format or define a new capacity between CD and DVD?
This should be the least effort compared to all the hassles with DSD.
Maybe this is the wrong thread as i own an ADI 2/4...
So here is my issue:
I run 2/4 with Pono and enabled, Ref Level +1dB, both trim Gains +6dB.
Input comes from an a Shure V15V-MR using XLR input with 100kOhm soldered between pin 1 and 2.
If amplification is set to +26dB auto stand-by works as expected. In this setting I can sometimes see a single bar showing up at the bottom of the meter caused by noise.
Setting amplification to +32dB (what I need to get full level) I see some more bars at the bottom of the meter and auto stand-by never gets activated. This means the ADI-2/4 stays on for forever.
In my opinion there should be an option to disable triggering auto stand-by analog input in the I/O settings.
What do you think? Anyone out the with the same issue?
Best DrCWO
The typical noise and hum of a phono input exceeds the threshold set for analog input Auto Standby detection. That is expected. I will check if we can set this threshold a bit higher without ill side effects.
RME User Forum → ADI-2 & 2/4 Pro series, ADI-2 DAC series → auto stand-by ADI-2 Pro
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