This is related to the increase in the dynamic range of the file (minimum to maximum amplitude) from 16bit to 24 or 32bit, which is unmasking harmonic energy that exists in the source file.
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Apologies, didnāt know it was iOS only. This is for Android and has a graphing function:
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That prompts a mal ware alert for me 
Damn. 
More searching to doā¦ 
Side effect with this increased dynamic: I can turn the volume knob of my stereo amplifier now by a ridiculous small angle only because otherwise it simply gets too loud. That kind of forced me to activate Audirvanaās Ā»Volume ControlĀ«, as a result, to specify a SW volume of -20dB. It works, but Iām not too happy with this method as such.
AudirvÄna employs 32bit digital Volume controlā¦
-20dB of attenuation = -3.3 bits reduction of dynamic range. ( 1 bit = 6dB )
You are not losing anything in this case. 
Theoretical Dynamic Range is calculated as:
[ 6 (db) x (number of bits) + 1.75mv = dynamic range in mv ]
However are you using plug-in DSP? Because you may be adding gain through EQ and this must be managed in the plug-in.
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Iām using MathAudioās Room-EQ. And as far as I understand its UI, I am rather reducing the volume there, with my current settings.
It is important to monitor the output level of the plug-inā¦
From the Room EQ Help Guideā¦
15. Adjust the position of the āRoom EQ gainā slider
Room EQ doesnāt clip the sound. However, the amplitude of its output signal can exceed 0 dB. If you use the volume control and the clip indicator of your host program, set the āRoom EQ gainā slider to 0 dB and donāt use it any more. Otherwise, play an audio file and look at the indicator which is shown in Fig.8. The red color of this indicator means that the amplitude of the Room EQ output signal exceeds 0 dB. Click the indicator to move the āRoom EQ gainā slider to the position corresponding to 0 dB output. Repeat this process with a few other audio files.
Fig.8. Indicator.
Other than the vertical slider in the top right half of the UI, thereās no way of directly adressing the output level that I knew of.
The Room EQ gain slider is meant to prevent clipping. I set that with the Ā»loudestĀ« tune I have in my sound library 
What is the impedance of your DAC output and what is the impedance of your amplifier inputā¦? are you using balanced XLR or single-ended RCA inputsā¦?
What happens when you lower the Room EQ gain by -20db and set the AudirvÄna output volume to 0dB? The plug-in is not expecting the increase in dynamic range because the up-sampling is done after the plug-in module.
All these are unchanged for years. Only change: I replaced the DACās original USB-A-to-USB-C cable with a USB-C-to-USB-C cable recently.
Minimum here is -6dB. Again: This slider is to prevent clipping only.
Did you make the room correction with up-sampling enabled?
Wellā¦ the test is to disable the Room EQ plug-in to understand if there is gain being applied in the plug-in, when the AudirvÄna Volume is set to 0dBā¦
A quick check showed no noteworthy difference regarding output volume.
Mathematically thatās correct, but in reality āmaximum upsamplingā can still sound better ā it does in my system. Probably depends on DAC. When I tested my receiver I tried both maximum and āpower of twoā, but the differences between receiver and Benchmark DAC (and the differences between DSD and PCM) were way bigger to my ears than the differences in PCM upsampling rates.
I now use DSD conversion, but my Benchmark DAC sounded ever-so-slightly more natural to my ears with 384kHz than with power-of-two. Possibly because the DACās internal processing works differently. Itās one thing if Audirvanaās math is optimalā¦ but the DACās further math (or conversion to intermediary bit rates) can apparently outweigh this theoretical advantage.
I started with power of two also because of the theoretical rounding errors and the āgeneral adviseā found on forums and SoX specifications. But 64 bit real number processing in todayās Macs and PCs possibly makes these errors minute. Mostly by chance, a few years ago I tried 384kHz for everything and, while it took a long time to hear (mostly feel) the difference, I found I enjoyed that more in my setup. I found the difference in what I heard very, very subtle though. I can rotate the acoustic treatment in my room by a few degrees or replace cables for a way bigger effectā¦
I would suspect that power-of-two is more relevant for 2x upsampling. To my ears the difference between any upsampling rates above 96kHz become extremely subtle. More subtle than tuning SoX parameters, or the conversion to DSD.
Iām still curious about what you and others hear though. Itās possible that other limitations in my system (or my ears) still mask the effect, even though Iām already super happy with the huge progress Iāve made over the past years, the latest Audirvana SoX tweaks and (since two weeks) Auva isolation feet having surprisingly big additional positive effects.
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Yep.
I will put it this way from my experience with my playback systemā¦
I will always choose the logical resultā¦ This is why I decimate DSD files to DXD 24/352.8kHz not 384kHzā¦ I have tuned my system for DSD playbackā¦ I never listen to PCMā¦
I donāt know that 64 bit math has anything to do with truncation/round-off error in a file, the issue there is primarily related to data accumulator/register overruns and under-runsā¦ The influence of round-off error energy is a function of the Fc.
Like @Jud says about modulating to 1-bit 22.4MHz, the potential for reducing noise is a good thingā¦ I donāt believe we need to go beyond 1-bit 5.6MHz in modulating PCM to DSD but now we are getting into some very subjective airā¦ The difference being in a juxtaposition to a native 1-bit 22.4MHz (DSD256) recording and a PCM recording being modulated to DSD256
My feeling that modulating to DSD256 or DSD512 and the appreciation of the resulting output sound-quality is DAC architecture dependentā¦ Interpolation slew-rate becomes more significant as the sample-frequency risesā¦
I presume that you mean that you still need to attenuate the output signal to your DAC? Are you adding gain in your DAC?
I up-sample all PCM to DSD128 with my AudirvÄna Volume disabledā¦ no overload and nominal output level to my HPAā¦